From b1d9e66c1c8d0e4ad29328c4f37266f292d722d3 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 3 May 2013 14:39:23 +0530 Subject: ASoC: 88pm860x: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 60159c07448d..1382f3f3f4bf 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1444,7 +1444,7 @@ static int pm860x_codec_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_IRQ, i); if (!res) { dev_err(&pdev->dev, "Failed to get IRQ resources\n"); - goto out; + return -EINVAL; } pm860x->irq[i] = res->start + chip->irq_base; strncpy(pm860x->name[i], res->name, MAX_NAME_LEN); @@ -1454,19 +1454,14 @@ static int pm860x_codec_probe(struct platform_device *pdev) pm860x_dai, ARRAY_SIZE(pm860x_dai)); if (ret) { dev_err(&pdev->dev, "Failed to register codec\n"); - goto out; + return -EINVAL; } return ret; - -out: - platform_set_drvdata(pdev, NULL); - return -EINVAL; } static int pm860x_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); - platform_set_drvdata(pdev, NULL); return 0; } -- cgit v1.2.1 From 6ab2b7b415441fa46357bef883e1ead086de1387 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 8 May 2013 14:15:35 +0100 Subject: ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls Add initial support for runtime tuning for the ADSP cores. This is achieved by exposing the coefficient configuration blocks as ALSA binary controls. The current code assumes that no controls on the DSP are volatile. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 458 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 3 + 2 files changed, 454 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b649c0b2..137830611928 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -215,6 +216,36 @@ static struct { [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff { + struct device *dev; + struct list_head ctl_list; + struct regmap *regmap; +}; + +struct wm_coeff_ctl { + const char *name; + struct snd_card *card; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int dirty:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -334,6 +365,181 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->dirty = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl, + const struct snd_kcontrol_new *kctl) +{ + int ret; + struct snd_kcontrol *kcontrol; + + kcontrol = snd_ctl_new1(kctl, wm_coeff); + ret = snd_ctl_add(ctl->card, kcontrol); + if (ret < 0) { + dev_err(wm_coeff->dev, "Failed to add %s: %d\n", + kctl->name, ret); + return ret; + } + ctl->kcontrol = kcontrol; + return 0; +} + +struct wmfw_ctl_work { + struct wm_coeff *wm_coeff; + struct wm_coeff_ctl *ctl; + struct work_struct work; +}; + +static int wmfw_add_ctl(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = wm_coeff_add_kcontrol(wm_coeff, + ctl, kcontrol); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + list_add(&ctl->list, &wm_coeff->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -547,7 +753,156 @@ out: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp) +static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled || ctl->dirty) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled) + continue; + if (ctl->dirty) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + ctl->dirty = 0; + } + } + + return 0; +} + +static void wm_adsp_ctl_work(struct work_struct *work) +{ + struct wmfw_ctl_work *ctl_work = container_of(work, + struct wmfw_ctl_work, + work); + + wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl); + kfree(ctl_work); +} + +static int wm_adsp_create_control(struct snd_soc_codec *codec, + const struct wm_adsp_alg_region *region) + +{ + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct wm_coeff_ctl *ctl; + struct wmfw_ctl_work *ctl_work; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + return -EINVAL; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + return 0; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->dirty = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->card = codec->card->snd_card; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); + if (!ctl_work) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + ctl_work->wm_coeff = dsp->wm_coeff; + ctl_work->ctl = ctl; + INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); + schedule_work(&ctl_work->work); + + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + +static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -730,7 +1085,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -738,7 +1102,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -758,7 +1131,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -766,7 +1148,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -774,7 +1165,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(®ion->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -986,6 +1386,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; @@ -1023,7 +1424,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1031,6 +1432,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -1047,6 +1458,11 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } break; default: @@ -1099,6 +1515,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; @@ -1164,7 +1581,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1172,6 +1589,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1209,6 +1636,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1247,36 +1679,48 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); + adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), + GFP_KERNEL); + if (!adsp->wm_coeff) + return -ENOMEM; + adsp->wm_coeff->regmap = adsp->regmap; + adsp->wm_coeff->dev = adsp->dev; + INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } } return 0; + +out_coeff: + kfree(adsp->wm_coeff); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index fea514627526..6e890b916592 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -55,6 +56,8 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct wm_coeff *wm_coeff; }; #define WM_ADSP1(wname, num) \ -- cgit v1.2.1 From 9b74fad508049471a8c783b9960b7834aba76293 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 May 2013 20:26:03 +0100 Subject: ASoC: sam9g20ek: Let device core handle pinctrl Since commit ab78029 (drivers/pinctrl: grab default handles from device core) we can rely on device core for handling pinctrl so remove devm_pinctrl_get_select_default() from the driver. Signed-off-by: Mark Brown Tested-by: Bo Shen Acked-by: Bo Shen --- sound/soc/atmel/sam9g20_wm8731.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 2d6fbd0125b9..802717eccbd0 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -38,8 +38,6 @@ #include #include -#include - #include #include @@ -203,15 +201,8 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) struct device_node *codec_np, *cpu_np; struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; - struct pinctrl *pinctrl; int ret; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "Failed to request pinctrl for mck\n"); - return PTR_ERR(pinctrl); - } - if (!np) { if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) -- cgit v1.2.1 From 10e8aa9af170886266618370e2ef330e0b2055bb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Micha=C5=82=20Miros=C5=82aw?= Date: Sat, 4 May 2013 22:21:38 +0200 Subject: ASoC: fix kernel message grepability MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Michał Mirosław Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 77 +++++++++++++++++++++++++++++----------------------- 1 file changed, 43 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d56bbea6e75e..308895a438d6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -272,8 +272,8 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) codec->debugfs_codec_root = debugfs_create_dir(codec->name, debugfs_card_root); if (!codec->debugfs_codec_root) { - dev_warn(codec->dev, "ASoC: Failed to create codec debugfs" - " directory\n"); + dev_warn(codec->dev, + "ASoC: Failed to create codec debugfs directory\n"); return; } @@ -286,8 +286,8 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) codec->debugfs_codec_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) - dev_warn(codec->dev, "ASoC: Failed to create codec register" - " debugfs file\n"); + dev_warn(codec->dev, + "ASoC: Failed to create codec register debugfs file\n"); snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); } @@ -631,8 +631,7 @@ int snd_soc_suspend(struct device *dev) */ if (codec->dapm.idle_bias_off) { dev_dbg(codec->dev, - "ASoC: idle_bias_off CODEC on" - " over suspend\n"); + "ASoC: idle_bias_off CODEC on over suspend\n"); break; } case SND_SOC_BIAS_OFF: @@ -643,8 +642,8 @@ int snd_soc_suspend(struct device *dev) regcache_mark_dirty(codec->control_data); break; default: - dev_dbg(codec->dev, "ASoC: CODEC is on" - " over suspend\n"); + dev_dbg(codec->dev, + "ASoC: CODEC is on over suspend\n"); break; } } @@ -713,8 +712,8 @@ static void soc_resume_deferred(struct work_struct *work) codec->suspended = 0; break; default: - dev_dbg(codec->dev, "ASoC: CODEC was on over" - " suspend\n"); + dev_dbg(codec->dev, + "ASoC: CODEC was on over suspend\n"); break; } } @@ -1110,8 +1109,8 @@ static int soc_probe_codec(struct snd_soc_card *card, } WARN(codec->dapm.idle_bias_off && codec->dapm.bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias" - " with idle_bias_off==1\n", codec->name); + "codec %s can not start from non-off bias with idle_bias_off==1\n", + codec->name); } /* If the driver didn't set I/O up try regmap */ @@ -1582,8 +1581,9 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, codec->compress_type = compress_type; ret = snd_soc_cache_init(codec); if (ret < 0) { - dev_err(codec->dev, "ASoC: Failed to set cache compression" - " type: %d\n", ret); + dev_err(codec->dev, + "ASoC: Failed to set cache compression type: %d\n", + ret); return ret; } codec->cache_init = 1; @@ -1639,8 +1639,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, card->owner, 0, &card->snd_card); if (ret < 0) { - dev_err(card->dev, "ASoC: can't create sound card for" - " card %s: %d\n", card->name, ret); + dev_err(card->dev, + "ASoC: can't create sound card for card %s: %d\n", + card->name, ret); goto base_error; } card->snd_card->dev = card->dev; @@ -1815,8 +1816,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) for (i = 0; i < card->num_rtd; i++) { ret = soc_register_ac97_dai_link(&card->rtd[i]); if (ret < 0) { - dev_err(card->dev, "ASoC: failed to register AC97:" - " %d\n", ret); + dev_err(card->dev, + "ASoC: failed to register AC97: %d\n", ret); while (--i >= 0) soc_unregister_ac97_dai_link(card->rtd[i].codec); goto probe_aux_dev_err; @@ -3586,14 +3587,16 @@ int snd_soc_register_card(struct snd_soc_card *card) * not both or neither. */ if (!!link->codec_name == !!link->codec_of_node) { - dev_err(card->dev, "ASoC: Neither/both codec" - " name/of_node are set for %s\n", link->name); + dev_err(card->dev, + "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); return -EINVAL; } /* Codec DAI name must be specified */ if (!link->codec_dai_name) { - dev_err(card->dev, "ASoC: codec_dai_name not" - " set for %s\n", link->name); + dev_err(card->dev, + "ASoC: codec_dai_name not set for %s\n", + link->name); return -EINVAL; } @@ -3602,8 +3605,9 @@ int snd_soc_register_card(struct snd_soc_card *card) * can be left unspecified, and a dummy platform will be used. */ if (link->platform_name && link->platform_of_node) { - dev_err(card->dev, "ASoC: Both platform name/of_node" - " are set for %s\n", link->name); + dev_err(card->dev, + "ASoC: Both platform name/of_node are set for %s\n", + link->name); return -EINVAL; } @@ -3613,8 +3617,9 @@ int snd_soc_register_card(struct snd_soc_card *card) * name alone.. */ if (link->cpu_name && link->cpu_of_node) { - dev_err(card->dev, "ASoC: Neither/both " - "cpu name/of_node are set for %s\n",link->name); + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); return -EINVAL; } /* @@ -3623,8 +3628,9 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (!link->cpu_dai_name && !(link->cpu_name || link->cpu_of_node)) { - dev_err(card->dev, "ASoC: Neither cpu_dai_name nor " - "cpu_name/of_node are set for %s\n", link->name); + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); return -EINVAL; } } @@ -3728,8 +3734,9 @@ static inline char *fmt_multiple_name(struct device *dev, struct snd_soc_dai_driver *dai_drv) { if (dai_drv->name == NULL) { - dev_err(dev, "ASoC: error - multiple DAI %s registered with" - " no name\n", dev_name(dev)); + dev_err(dev, + "ASoC: error - multiple DAI %s registered with no name\n", + dev_name(dev)); return NULL; } @@ -3859,8 +3866,9 @@ static int snd_soc_register_dais(struct device *dev, list_for_each_entry(codec, &codec_list, list) { if (codec->dev == dev) { - dev_dbg(dev, "ASoC: Mapped DAI %s to " - "CODEC %s\n", dai->name, codec->name); + dev_dbg(dev, + "ASoC: Mapped DAI %s to CODEC %s\n", + dai->name, codec->name); dai->codec = codec; break; } @@ -4296,8 +4304,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, num_routes = of_property_count_strings(np, propname); if (num_routes < 0 || num_routes & 1) { - dev_err(card->dev, "ASoC: Property '%s' does not exist or its" - " length is not even\n", propname); + dev_err(card->dev, + "ASoC: Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; -- cgit v1.2.1 From 8011412999484a82a23dc3c9a5c9d5a1677ca05d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 25 Feb 2013 15:14:19 +0000 Subject: ASoC: dapm: Provide early event callbacks for power up and down Some devices may benefit from being able to start some parts of the widget power up/down sequence earlier on in the sequence than the point at which the final power state is committed. Support these by providing events which are called before any power state changes are done. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a80c883bb8be..e4e5420de725 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1277,6 +1277,14 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, ev_name = "POST_PMD"; power = 0; break; + case SND_SOC_DAPM_WILL_PMU: + ev_name = "WILL_PMU"; + power = 1; + break; + case SND_SOC_DAPM_WILL_PMD: + ev_name = "WILL_PMD"; + power = 0; + break; default: BUG(); return; @@ -1737,6 +1745,14 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); + list_for_each_entry(w, &down_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); + } + + list_for_each_entry(w, &up_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); + } + /* Power down widgets first; try to avoid amplifying pops. */ dapm_seq_run(dapm, &down_list, event, false); -- cgit v1.2.1 From 701c73aa89032f2551cdc73725cb881c0886d86f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 3 May 2013 14:39:22 +0530 Subject: ASoC: ep93xx: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 7798fbd5e81d..840c9b18201e 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -403,7 +403,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return 0; fail: - platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; dev_set_drvdata(&pdev->dev, NULL); return ret; @@ -418,7 +417,6 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) /* disable the AC97 controller */ ep93xx_ac97_write_reg(info, AC97GCR, 0); - platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; dev_set_drvdata(&pdev->dev, NULL); -- cgit v1.2.1 From 2fc059f2cc875a6d7372057093cd78cc9284b555 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 24 Apr 2013 11:54:43 -0300 Subject: ASoC: imx-sgtl5000: Do not enter the error path on success Return on success instead of entering the error path. Signed-off-by: Fabio Estevam Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78858df..5a6aaa3b947a 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -174,6 +174,11 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + clk_fail: clk_put(data->codec_clk); fail: -- cgit v1.2.1 From f8b24fcbd05bf7a9112812bf6ec60679ae928801 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 24 Apr 2013 13:23:14 -0300 Subject: ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable Variable 'ret' is not needed here, so just remove it. Signed-off-by: Fabio Estevam Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b1d9b5ebeeeb..4f74b051f73a 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -116,7 +116,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i, ret = 0; + int i; if (!np) return 1; /* no device tree */ @@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return ret; + return 0; } static int mxs_sgtl5000_probe(struct platform_device *pdev) -- cgit v1.2.1 From 666c25e3d759dfd33c6df4cd3a26f1bfed65215e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 24 Apr 2013 13:23:15 -0300 Subject: ASoC: mxs-sgtl5000: Remove unneeded fields from snd_soc_dai_link It makes no sense to hardcode the I2C bus into the codec_name field. cpu_dai_name and platform_name are also overwritten later in mxs_sgtl5000_probe_dt(). So remove the three fields, as mxs platform is dt-only platform. Signed-off-by: Fabio Estevam Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 4f74b051f73a..1b134d72f120 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -90,17 +90,11 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .name = "HiFi Tx", .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", - .codec_name = "sgtl5000.0-000a", - .cpu_dai_name = "mxs-saif.0", - .platform_name = "mxs-saif.0", .ops = &mxs_sgtl5000_hifi_ops, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", - .codec_name = "sgtl5000.0-000a", - .cpu_dai_name = "mxs-saif.1", - .platform_name = "mxs-saif.1", .ops = &mxs_sgtl5000_hifi_ops, }, }; -- cgit v1.2.1 From 436947fc82237e2cd78b3b2c11633aaa6ef07641 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 6 May 2013 17:39:08 -0300 Subject: ASoC: fsl: imx-audmux: Let device core handle pinctrl Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 47f046a8fdab..e260f1f899db 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -26,7 +26,6 @@ #include #include #include -#include #include "imx-audmux.h" @@ -247,7 +246,6 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; - struct pinctrl *pinctrl; const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); @@ -256,12 +254,6 @@ static int imx_audmux_probe(struct platform_device *pdev) if (IS_ERR(audmux_base)) return PTR_ERR(audmux_base); - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "setup pinctrl failed!"); - return PTR_ERR(pinctrl); - } - audmux_clk = devm_clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", -- cgit v1.2.1 From bc2716fb5a3bb05abc81c5c4c367cca99854fb9b Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 3 May 2013 14:39:24 +0530 Subject: ASoC: jz4740: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/jz4740.c | 2 -- sound/soc/jz4740/jz4740-i2s.c | 1 - 2 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 5f607b35b68b..bcebd1a9ce31 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -384,8 +384,6 @@ static int jz4740_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); - platform_set_drvdata(pdev, NULL); - return 0; } diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9a126441c5f3..cafc6eda0ac5 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -509,7 +509,6 @@ static int jz4740_i2s_dev_remove(struct platform_device *pdev) iounmap(i2s->base); release_mem_region(i2s->mem->start, resource_size(i2s->mem)); - platform_set_drvdata(pdev, NULL); kfree(i2s); return 0; -- cgit v1.2.1 From e76af6d189075a0ca59bc654157e80da53559fb0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 6 May 2013 15:06:01 -0300 Subject: ASoC: mxs: mxs-saif: Let device core handle pinctrl Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index d31dc52fa862..71a972f5af97 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include @@ -667,7 +666,6 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct pinctrl *pinctrl; int ret = 0; struct device_node *master; @@ -707,12 +705,6 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[saif->id] = saif; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - ret = PTR_ERR(pinctrl); - return ret; - } - saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); -- cgit v1.2.1 From 3d77876b8fd43ba845f3503369590a9bcc1f1e0c Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 3 May 2013 14:39:20 +0530 Subject: ASoC: omap-mcbsp: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index eadbfb6b5000..7483efb6dc67 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -814,8 +814,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) clk_put(mcbsp->fclk); - platform_set_drvdata(pdev, NULL); - return 0; } -- cgit v1.2.1 From e75fa9b1f9d430edeb848db329f924996bc964d6 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 3 May 2013 14:39:19 +0530 Subject: ASoC: Samsung: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index e43bd4294f99..23a9204b106d 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -176,7 +176,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 3688a32000a2..0c84ca099612 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -146,7 +146,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } -- cgit v1.2.1 From 271f193d739f894a7d1229c2cbb72d2f8b847544 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 3 May 2013 18:04:24 -0300 Subject: ASoC: sgtl5000: Fix comment about register addresses The list below the comment relates to sgtl5000 registers addresses, so change the comment to improve the description. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 8a9f43534b79..4b69229a9818 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -12,7 +12,7 @@ #define _SGTL5000_H /* - * Register values. + * Registers addresses */ #define SGTL5000_CHIP_ID 0x0000 #define SGTL5000_CHIP_DIG_POWER 0x0002 -- cgit v1.2.1 From e5d80e82e32e4cb66d67a56352e5a594e2a35cd0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 4 May 2013 15:39:34 -0300 Subject: ASoC: sgtl5000: Convert to use regmap directly Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 145 +++++++++++++++++++++++++++++++++----------- 1 file changed, 111 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec9b107..327b4434b4ce 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -34,30 +35,30 @@ #define SGTL5000_MAX_REG_OFFSET 0x013A /* default value of sgtl5000 registers */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { - [SGTL5000_CHIP_CLK_CTRL] = 0x0008, - [SGTL5000_CHIP_I2S_CTRL] = 0x0010, - [SGTL5000_CHIP_SSS_CTRL] = 0x0008, - [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, - [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, - [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, - [SGTL5000_CHIP_ANA_CTRL] = 0x0111, - [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, - [SGTL5000_CHIP_ANA_POWER] = 0x7060, - [SGTL5000_CHIP_PLL_CTRL] = 0x5000, - [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, - [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, - [SGTL5000_DAP_SURROUND] = 0x0040, - [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, - [SGTL5000_DAP_MAIN_CHAN] = 0x8000, - [SGTL5000_DAP_AVC_CTRL] = 0x0510, - [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, - [SGTL5000_DAP_AVC_ATTACK] = 0x0028, - [SGTL5000_DAP_AVC_DECAY] = 0x0050, +static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, + { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, + { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, + { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, + { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, + { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, + { SGTL5000_CHIP_ANA_POWER, 0x7060 }, + { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, + { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_SURROUND, 0x0040 }, + { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, + { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, + { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, + { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -112,6 +113,7 @@ struct sgtl5000_priv { int fmt; /* i2s data format */ struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; + struct regmap *regmap; }; /* @@ -958,17 +960,76 @@ static struct snd_soc_dai_driver sgtl5000_dai = { .symmetric_rates = 1, }; -static int sgtl5000_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool sgtl5000_volatile(struct device *dev, unsigned int reg) { switch (reg) { case SGTL5000_CHIP_ID: case SGTL5000_CHIP_ADCDAC_CTRL: case SGTL5000_CHIP_ANA_STATUS: - return 1; + return true; } - return 0; + return false; +} + +static bool sgtl5000_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_DIG_POWER: + case SGTL5000_CHIP_CLK_CTRL: + case SGTL5000_CHIP_I2S_CTRL: + case SGTL5000_CHIP_SSS_CTRL: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_DAC_VOL: + case SGTL5000_CHIP_PAD_STRENGTH: + case SGTL5000_CHIP_ANA_ADC_CTRL: + case SGTL5000_CHIP_ANA_HP_CTRL: + case SGTL5000_CHIP_ANA_CTRL: + case SGTL5000_CHIP_LINREG_CTRL: + case SGTL5000_CHIP_REF_CTRL: + case SGTL5000_CHIP_MIC_CTRL: + case SGTL5000_CHIP_LINE_OUT_CTRL: + case SGTL5000_CHIP_LINE_OUT_VOL: + case SGTL5000_CHIP_ANA_POWER: + case SGTL5000_CHIP_PLL_CTRL: + case SGTL5000_CHIP_CLK_TOP_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + case SGTL5000_CHIP_SHORT_CTRL: + case SGTL5000_CHIP_ANA_TEST2: + case SGTL5000_DAP_CTRL: + case SGTL5000_DAP_PEQ: + case SGTL5000_DAP_BASS_ENHANCE: + case SGTL5000_DAP_BASS_ENHANCE_CTRL: + case SGTL5000_DAP_AUDIO_EQ: + case SGTL5000_DAP_SURROUND: + case SGTL5000_DAP_FLT_COEF_ACCESS: + case SGTL5000_DAP_COEF_WR_B0_MSB: + case SGTL5000_DAP_COEF_WR_B0_LSB: + case SGTL5000_DAP_EQ_BASS_BAND0: + case SGTL5000_DAP_EQ_BASS_BAND1: + case SGTL5000_DAP_EQ_BASS_BAND2: + case SGTL5000_DAP_EQ_BASS_BAND3: + case SGTL5000_DAP_EQ_BASS_BAND4: + case SGTL5000_DAP_MAIN_CHAN: + case SGTL5000_DAP_MIX_CHAN: + case SGTL5000_DAP_AVC_CTRL: + case SGTL5000_DAP_AVC_THRESHOLD: + case SGTL5000_DAP_AVC_ATTACK: + case SGTL5000_DAP_AVC_DECAY: + case SGTL5000_DAP_COEF_WR_B1_MSB: + case SGTL5000_DAP_COEF_WR_B1_LSB: + case SGTL5000_DAP_COEF_WR_B2_MSB: + case SGTL5000_DAP_COEF_WR_B2_LSB: + case SGTL5000_DAP_COEF_WR_A1_MSB: + case SGTL5000_DAP_COEF_WR_A1_LSB: + case SGTL5000_DAP_COEF_WR_A2_MSB: + case SGTL5000_DAP_COEF_WR_A2_LSB: + return true; + + default: + return false; + } } #ifdef CONFIG_SUSPEND @@ -1300,7 +1361,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* setup i2c data ops */ - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + codec->control_data = sgtl5000->regmap; + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1391,11 +1453,6 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .suspend = sgtl5000_suspend, .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = sgtl5000_regs, - .volatile_register = sgtl5000_volatile_register, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), .dapm_widgets = sgtl5000_dapm_widgets, @@ -1404,6 +1461,19 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes), }; +static const struct regmap_config sgtl5000_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = SGTL5000_MAX_REG_OFFSET, + .volatile_reg = sgtl5000_volatile, + .readable_reg = sgtl5000_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sgtl5000_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), +}; + static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -1415,6 +1485,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (!sgtl5000) return -ENOMEM; + sgtl5000->regmap = devm_regmap_init_i2c(client, &sgtl5000_regmap); + if (IS_ERR(sgtl5000->regmap)) { + ret = PTR_ERR(sgtl5000->regmap); + dev_err(&client->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, -- cgit v1.2.1 From 29f421c2107f28976dacc9f1fca4bbfebee5b10f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 24 Apr 2013 13:23:14 -0300 Subject: ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable Variable 'ret' is not needed here, so just remove it. Signed-off-by: Fabio Estevam Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b1d9b5ebeeeb..4f74b051f73a 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -116,7 +116,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i, ret = 0; + int i; if (!np) return 1; /* no device tree */ @@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return ret; + return 0; } static int mxs_sgtl5000_probe(struct platform_device *pdev) -- cgit v1.2.1 From 24279dcee5456bb141a1ca007a3f0abe02bf91d0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 24 Apr 2013 13:23:15 -0300 Subject: ASoC: mxs-sgtl5000: Remove unneeded fields from snd_soc_dai_link It makes no sense to hardcode the I2C bus into the codec_name field. cpu_dai_name and platform_name are also overwritten later in mxs_sgtl5000_probe_dt(). So remove the three fields, as mxs platform is dt-only platform. Signed-off-by: Fabio Estevam Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 4f74b051f73a..1b134d72f120 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -90,17 +90,11 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .name = "HiFi Tx", .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", - .codec_name = "sgtl5000.0-000a", - .cpu_dai_name = "mxs-saif.0", - .platform_name = "mxs-saif.0", .ops = &mxs_sgtl5000_hifi_ops, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", - .codec_name = "sgtl5000.0-000a", - .cpu_dai_name = "mxs-saif.1", - .platform_name = "mxs-saif.1", .ops = &mxs_sgtl5000_hifi_ops, }, }; -- cgit v1.2.1 From b871f1ad3c8a1ac2fb862f9261f14a67dc2c7b7d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 9 May 2013 21:15:46 -0300 Subject: ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe() The usual place for reading chip ID is inside i2c_probe, so move it there and also convert it to regmap. sgtl5000_enable_regulators() needs to read the chip revision, so keep the revision check there. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 39 ++++++++++++++++++++++++--------------- 1 file changed, 24 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b4434b4ce..1ab356ab0f31 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1275,7 +1275,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - u16 reg; + int reg; int ret; int rev; int i; @@ -1303,23 +1303,17 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* read chip information */ - reg = snd_soc_read(codec, SGTL5000_CHIP_ID); - if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != - SGTL5000_PARTID_PART_ID) { - dev_err(codec->dev, - "Device with ID register %x is not a sgtl5000\n", reg); - ret = -ENODEV; - goto err_regulator_disable; - } - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); - /* * workaround for revision 0x11 and later, * roll back to use internal LDO */ + + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); + if (ret) + goto err_regulator_disable; + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + if (external_vddd && rev >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), @@ -1478,7 +1472,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret; + int ret, reg, rev; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1492,6 +1486,21 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + /* read chip information */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); + if (ret) + return ret; + + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(&client->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + return -ENODEV; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, -- cgit v1.2.1 From af8ee11209e749c75eabf32b2a4ca631f396acf8 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 9 May 2013 21:15:47 -0300 Subject: ASoC: sgtl5000: Fix driver probe after reset After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) sgtl5000 codec does not have a reset line, nor a reset command in software, so after a system reset the codec does not contain the default register values from sgtl5000_reg_defaults[] anymore, as these are only valid after a power-on-reset cycle. Fix this issue by explicitly reading all the reset register values from sgtl5000_reg_defaults[] and writing them back into sgtl5000 to ensure a sane state. Signed-off-by: Fabio Estevam Tested-by: Eric Nelson Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1ab356ab0f31..1c3b20fc7ec3 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1468,6 +1468,31 @@ static const struct regmap_config sgtl5000_regmap = { .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), }; +/* + * Write all the default values from sgtl5000_reg_defaults[] array into the + * sgtl5000 registers, to make sure we always start with the sane registers + * values as stated in the datasheet. + * + * Since sgtl5000 does not have a reset line, nor a reset command in software, + * we follow this approach to guarantee we always start from the default values + * and avoid problems like, not being able to probe after an audio playback + * followed by a system reset or a 'reboot' command in Linux + */ +static int sgtl5000_fill_defaults(struct sgtl5000_priv *sgtl5000) +{ + int i, ret, val, index; + + for (i = 0; i < ARRAY_SIZE(sgtl5000_reg_defaults); i++) { + val = sgtl5000_reg_defaults[i].def; + index = sgtl5000_reg_defaults[i].reg; + ret = regmap_write(sgtl5000->regmap, index, val); + if (ret) + return ret; + } + + return 0; +} + static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -1503,6 +1528,11 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, sgtl5000); + /* Ensure sgtl5000 will start with sane register values */ + ret = sgtl5000_fill_defaults(sgtl5000); + if (ret) + return ret; + ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); return ret; -- cgit v1.2.1 From ee492cfcb17631e4345fa97f205ca9617fffaebc Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Thu, 25 Apr 2013 15:13:12 +0200 Subject: ASoC: spdif_transceiver: Change driver filename to spdif_transmitter.c. Transceiver usually means receiver + transmitter. This codec can do only transmit. Update driver accordingly. Signed-off-by: Marek Belisko Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/spdif_transciever.c | 69 ------------------------------------ sound/soc/codecs/spdif_transmitter.c | 69 ++++++++++++++++++++++++++++++++++++ 3 files changed, 70 insertions(+), 70 deletions(-) delete mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transmitter.c (limited to 'sound') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9a1f4c..dae0aa60dac0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -50,7 +50,7 @@ snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c deleted file mode 100644 index 112a49d66e39..000000000000 --- a/sound/soc/codecs/spdif_transciever.c +++ /dev/null @@ -1,69 +0,0 @@ -/* - * ALSA SoC SPDIF DIT driver - * - * This driver is used by controllers which can operate in DIT (SPDI/F) where - * no codec is needed. This file provides stub codec that can be used - * in these configurations. TI DaVinci Audio controller uses this driver. - * - * Author: Steve Chen, - * Copyright: (C) 2009 MontaVista Software, Inc., - * Copyright: (C) 2009 Texas Instruments, India - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include - -#define DRV_NAME "spdif-dit" - -#define STUB_RATES SNDRV_PCM_RATE_8000_96000 -#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE - - -static struct snd_soc_codec_driver soc_codec_spdif_dit; - -static struct snd_soc_dai_driver dit_stub_dai = { - .name = "dit-hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 384, - .rates = STUB_RATES, - .formats = STUB_FORMATS, - }, -}; - -static int spdif_dit_probe(struct platform_device *pdev) -{ - return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dit, - &dit_stub_dai, 1); -} - -static int spdif_dit_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); - return 0; -} - -static struct platform_driver spdif_dit_driver = { - .probe = spdif_dit_probe, - .remove = spdif_dit_remove, - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(spdif_dit_driver); - -MODULE_AUTHOR("Steve Chen "); -MODULE_DESCRIPTION("SPDIF dummy codec driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c new file mode 100644 index 000000000000..112a49d66e39 --- /dev/null +++ b/sound/soc/codecs/spdif_transmitter.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIT (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. TI DaVinci Audio controller uses this driver. + * + * Author: Steve Chen, + * Copyright: (C) 2009 MontaVista Software, Inc., + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#define DRV_NAME "spdif-dit" + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + + +static struct snd_soc_codec_driver soc_codec_spdif_dit; + +static struct snd_soc_dai_driver dit_stub_dai = { + .name = "dit-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dit, + &dit_stub_dai, 1); +} + +static int spdif_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver spdif_dit_driver = { + .probe = spdif_dit_probe, + .remove = spdif_dit_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(spdif_dit_driver); + +MODULE_AUTHOR("Steve Chen "); +MODULE_DESCRIPTION("SPDIF dummy codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.1 From 1b7c8b350fd4751051f0abba040a29b72f829665 Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Thu, 25 Apr 2013 15:13:13 +0200 Subject: ASoC: spdif_transmitter: Add DT support. Add devicetree support for this dummy audio soc driver. Signed-off-by: Michal Bachraty Signed-off-by: Marek Belisko Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transmitter.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 112a49d66e39..18280499fd55 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -20,6 +20,7 @@ #include #include #include +#include #define DRV_NAME "spdif-dit" @@ -52,12 +53,21 @@ static int spdif_dit_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dit_dt_ids[] = { + { .compatible = "linux,spdif-dit", }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dit_dt_ids); +#endif + static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dit_dt_ids), }, }; -- cgit v1.2.1 From f9c8ba8965597bb45b95014338d59ade15d53e93 Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Thu, 25 Apr 2013 15:13:14 +0200 Subject: ASoC: spdif_receiver: Add DT support. Add devicetree support for this dummy audio soc driver. Signed-off-by: Michal Bachraty Signed-off-by: Marek Belisko Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_receiver.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index dd8d856053fc..e9d7881ed2c8 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -21,6 +21,7 @@ #include #include #include +#include #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -51,12 +52,21 @@ static int spdif_dir_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dir_dt_ids[] = { + { .compatible = "linux,spdif-dir", }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dir_dt_ids); +#endif + static struct platform_driver spdif_dir_driver = { .probe = spdif_dir_probe, .remove = spdif_dir_remove, .driver = { .name = "spdif-dir", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dir_dt_ids), }, }; -- cgit v1.2.1 From 46fdd8b11d4ab58af126344dcbc0bd565174db16 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:06 +0200 Subject: ASoC: spear: Setup dma data in DAI probe This allows us to access the DAI DMA data when we create the PCM. We'll use this when converting spear to generic DMA engine PCM driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/spear/spdif_in.c | 12 ++++-------- sound/soc/spear/spdif_out.c | 7 ++++--- 2 files changed, 8 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 14d57e89bcba..82c838753c06 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -49,15 +49,12 @@ static void spdif_in_configure(struct spdif_in_dev *host) writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); } -static int spdif_in_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int spdif_in_dai_probe(struct snd_soc_dai *dai) { - struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); - if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) - return -EINVAL; + dai->capture_dma_data = &host->dma_params; - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); return 0; } @@ -70,7 +67,6 @@ static void spdif_in_shutdown(struct snd_pcm_substream *substream, return; writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); - snd_soc_dai_set_dma_data(dai, substream, NULL); } static void spdif_in_format(struct spdif_in_dev *host, u32 format) @@ -151,13 +147,13 @@ static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, } static struct snd_soc_dai_ops spdif_in_dai_ops = { - .startup = spdif_in_startup, .shutdown = spdif_in_shutdown, .trigger = spdif_in_trigger, .hw_params = spdif_in_hw_params, }; struct snd_soc_dai_driver spdif_in_dai = { + .probe = spdif_in_dai_probe, .capture = { .channels_min = 2, .channels_max = 2, diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 1e3c3dda3598..12b4f2fcb9af 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -62,8 +62,6 @@ static int spdif_out_startup(struct snd_pcm_substream *substream, if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -EINVAL; - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); - ret = clk_enable(host->clk); if (ret) return ret; @@ -84,7 +82,6 @@ static void spdif_out_shutdown(struct snd_pcm_substream *substream, clk_disable(host->clk); host->running = false; - snd_soc_dai_set_dma_data(dai, substream, NULL); } static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq, @@ -245,6 +242,10 @@ static const struct snd_kcontrol_new spdif_out_controls[] = { int spdif_soc_dai_probe(struct snd_soc_dai *dai) { + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &host->dma_params; + return snd_soc_add_dai_controls(dai, spdif_out_controls, ARRAY_SIZE(spdif_out_controls)); } -- cgit v1.2.1 From 52c102e534fd46a25aacab37bbaaa593929a2ca1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:07 +0200 Subject: ASoC: spear: Use generic dmaengine PCM Use the generic dmaengine PCM driver instead of a custom implementation. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/spear/spear_pcm.c | 152 +++++--------------------------------------- 1 file changed, 15 insertions(+), 137 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 2fbd4899d8ef..4707f2b862c3 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -13,19 +13,13 @@ #include #include -#include -#include #include -#include -#include -#include #include #include -#include #include #include -static struct snd_pcm_hardware spear_pcm_hardware = { +static const struct snd_pcm_hardware spear_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), @@ -37,149 +31,33 @@ static struct snd_pcm_hardware spear_pcm_hardware = { .fifo_size = 0, /* fifo size in bytes */ }; -static int spear_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static struct dma_chan *spear_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) { - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + struct spear_dma_data *dma_data; - return 0; -} - -static int spear_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - - return 0; -} - -static int spear_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - struct spear_dma_data *dma_data = (struct spear_dma_data *) - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - int ret; - - ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware); - if (ret) - return ret; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - return snd_dmaengine_pcm_open_request_chan(substream, dma_data->filter, - dma_data); + return snd_dmaengine_pcm_request_channel(dma_data->filter, dma_data); } -static int spear_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops spear_pcm_ops = { - .open = spear_pcm_open, - .close = snd_dmaengine_pcm_close_release_chan, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = spear_pcm_hw_params, - .hw_free = spear_pcm_hw_free, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, - .mmap = spear_pcm_mmap, -}; - -static int -spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, - size_t size) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - - dev_info(buf->dev.dev, - " preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *)buf->area, (void *)buf->addr, size); - - buf->bytes = size; - return 0; -} - -static void spear_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf || !buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); - -static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - int ret; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &spear_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (rtd->cpu_dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, - SNDRV_PCM_STREAM_PLAYBACK, - spear_pcm_hardware.buffer_bytes_max); - if (ret) - return ret; - } - - if (rtd->cpu_dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, - SNDRV_PCM_STREAM_CAPTURE, - spear_pcm_hardware.buffer_bytes_max); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver spear_soc_platform = { - .ops = &spear_pcm_ops, - .pcm_new = spear_pcm_new, - .pcm_free = spear_pcm_free, +static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { + .pcm_hardware = &spear_pcm_hardware, + .compat_request_channel = spear_pcm_request_chan, + .prealloc_buffer_size = 16 * 1024, }; static int spear_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &spear_soc_platform); + return snd_dmaengine_pcm_register(&pdev->dev, + &spear_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT); } static int spear_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_platform(&pdev->dev); - + snd_dmaengine_pcm_unregister(&pdev->dev); return 0; } -- cgit v1.2.1 From bfcc74e6101a978b3987e767815595e5a9fec2ca Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 8 May 2013 11:47:38 +0200 Subject: ASoC: SPEAr spdif_{in,out}: use devm for clk and a few more cleanups MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Drop dev_set_drvdata as this is handled in the core and use devm_request_and_ioremap instead of devm_ioremap to properly register the address range used Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/spear/spdif_in.c | 7 +------ sound/soc/spear/spdif_out.c | 7 ++----- 2 files changed, 3 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 82c838753c06..643ada6a1fa5 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -231,7 +231,7 @@ static int spdif_in_probe(struct platform_device *pdev) if (host->irq < 0) return -EINVAL; - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -253,7 +253,6 @@ static int spdif_in_probe(struct platform_device *pdev) ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0, "spdif-in", host); if (ret) { - clk_put(host->clk); dev_warn(&pdev->dev, "request_irq failed\n"); return ret; } @@ -273,14 +272,10 @@ static int spdif_in_remove(struct platform_device *pdev) struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } - static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, .remove = spdif_in_remove, diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 12b4f2fcb9af..fd25cc5efe27 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -298,14 +298,14 @@ static int spdif_out_probe(struct platform_device *pdev) return -ENOMEM; } - host->io_base = devm_ioremap(&pdev->dev, res->start, + host->io_base = devm_request_and_ioremap(&pdev->dev, res->start, resource_size(res)); if (!host->io_base) { dev_warn(&pdev->dev, "ioremap failed\n"); return -ENOMEM; } - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -334,9 +334,6 @@ static int spdif_out_remove(struct platform_device *pdev) struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } -- cgit v1.2.1 From f656df65743451d77e30e44e014b301721dff7cf Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Tue, 30 Apr 2013 16:09:54 +0200 Subject: ASoC: ux500: register controls to card instead of codec Update mop500_ab8500_machine_init to register mop500_ab8500_ctrls as card control instead of codec control, as it only contains SOC_DAPM_PIN_SWITCH definitions. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 892ad9a05c9f..6a33788055f3 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -125,9 +125,9 @@ static int mop500_ab8500_set_mclk(struct device *dev, static int mclk_input_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct mop500_ab8500_drvdata *drvdata = - snd_soc_card_get_drvdata(codec->card); + snd_soc_card_get_drvdata(card); ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; @@ -137,9 +137,9 @@ static int mclk_input_control_get(struct snd_kcontrol *kcontrol, static int mclk_input_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct mop500_ab8500_drvdata *drvdata = - snd_soc_card_get_drvdata(codec->card); + snd_soc_card_get_drvdata(card); unsigned int val = ucontrol->value.enumerated.item[0]; if (val > (unsigned int)MCLK_ULPCLK) @@ -385,7 +385,7 @@ int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) drvdata->mclk_sel = MCLK_ULPCLK; /* Add controls */ - ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ret = snd_soc_add_card_controls(codec->card, mop500_ab8500_ctrls, ARRAY_SIZE(mop500_ab8500_ctrls)); if (ret < 0) { pr_err("%s: Failed to add machine-controls (%d)!\n", -- cgit v1.2.1 From 2e8e3880a15efacd21d68f77546ccd09f5e99521 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Thu, 2 May 2013 11:52:51 +0200 Subject: ASoC: ux500: drop clock gating widgets from machine driver Drop ab8500 clock gating widgets from mop500_ab8500_ctrls, as these bits are already controlled by ab8500 codec driver and should not be exposed to the user. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 6a33788055f3..44e25a291044 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -160,16 +160,6 @@ static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { SOC_ENUM_EXT("Master Clock Select", soc_enum_mclk, mclk_input_control_get, mclk_input_control_put), - /* Digital interface - Clocks */ - SOC_SINGLE("Digital Interface Master Generator Switch", - AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, - 1, 0), - SOC_SINGLE("Digital Interface 0 Bit-clock Switch", - AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, - 1, 0), - SOC_SINGLE("Digital Interface 1 Bit-clock Switch", - AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, - 1, 0), SOC_DAPM_PIN_SWITCH("Headset Left"), SOC_DAPM_PIN_SWITCH("Headset Right"), SOC_DAPM_PIN_SWITCH("Earpiece"), -- cgit v1.2.1 From b9600b4b1cf8b8f06b6a5d025eff160f41950485 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Wed, 8 May 2013 09:14:16 +0200 Subject: ASoC: ab8500-codec: Add missing ad_to_slot definitions According to the AB8500 user manual AD to Slot register multiplexer accept values from 0 to 15 where: 0 to 7 corresponds to AD_OUTx slots 8 to 11 corresponds to zero output 12 to 15 sets the output in tristate mode Update enum_ad_to_slot_map array to reflect this definition. This also allows alsamixer to properly display the default configuration, as all controls are set to tristate (=12) at reset. Signed-off-by: Fabio Baltieri Acked-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index a153b168129b..3126cac7b7c8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1496,6 +1496,12 @@ static const char * const enum_ad_to_slot_map[] = {"AD_OUT1", "AD_OUT7", "AD_OUT8", "zeroes", + "zeroes", + "zeroes", + "zeroes", + "tristate", + "tristate", + "tristate", "tristate"}; static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map, AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT, -- cgit v1.2.1 From 48dcf1d82fd8158ac8a9b843abb4965c69a19781 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 3 May 2013 14:39:21 +0530 Subject: ASoC: mid-x86: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 4139116c33b5..78d582519891 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -425,7 +425,6 @@ static int snd_mfld_mc_remove(struct platform_device *pdev) free_irq(platform_get_irq(pdev, 0), mc_drv_ctx); snd_soc_unregister_card(&snd_soc_card_mfld); kfree(mc_drv_ctx); - platform_set_drvdata(pdev, NULL); return 0; } -- cgit v1.2.1 From 010187fb45368470177b4d42916856f22a08a824 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 12 May 2013 20:07:39 +0200 Subject: ASoC: jz4740-i2s: Use clk_prepare_enable/clk_disable_unprepare In preparation to switching the jz4740 clk driver to the common clk framework update the clk enable/disable calls to clk_prepare_enable/clk_disable_unprepare. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index cafc6eda0ac5..4c849a49c72a 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -118,7 +118,7 @@ static int jz4740_i2s_startup(struct snd_pcm_substream *substream, ctrl |= JZ_AIC_CTRL_FLUSH; jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -140,7 +140,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, @@ -314,10 +314,10 @@ static int jz4740_i2s_suspend(struct snd_soc_dai *dai) conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } @@ -327,10 +327,10 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); if (dai->active) { - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -368,7 +368,7 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); jz4740_i2c_init_pcm_config(i2s); @@ -388,7 +388,7 @@ static int jz4740_i2s_dai_remove(struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } -- cgit v1.2.1 From d1a0a2995855e8d583c5cf97dbf0f6b376668c45 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 May 2013 21:40:10 +0100 Subject: ASoC: wm8994: Support EFS mode for FLL Later WM8994 devices support an enhanced accuracy FLL divisor mode called EFS which allows more precise selection of fractional source to output ratios. Support this on relevant devices. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 49 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 36 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1eb152cb1097..9f32dd8660d5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -2005,15 +2006,16 @@ struct fll_div { u16 outdiv; u16 n; u16 k; + u16 lambda; u16 clk_ref_div; u16 fll_fratio; }; -static int wm8994_get_fll_config(struct fll_div *fll, +static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, int freq_in, int freq_out) { u64 Kpart; - unsigned int K, Ndiv, Nmod; + unsigned int K, Ndiv, Nmod, gcd_fll; pr_debug("FLL input=%dHz, output=%dHz\n", freq_in, freq_out); @@ -2062,20 +2064,30 @@ static int wm8994_get_fll_config(struct fll_div *fll, Nmod = freq_out % freq_in; pr_debug("Nmod=%d\n", Nmod); - /* Calculate fractional part - scale up so we can round. */ - Kpart = FIXED_FLL_SIZE * (long long)Nmod; + switch (control->type) { + case WM8994: + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; - do_div(Kpart, freq_in); + do_div(Kpart, freq_in); - K = Kpart & 0xFFFFFFFF; + K = Kpart & 0xFFFFFFFF; - if ((K % 10) >= 5) - K += 5; + if ((K % 10) >= 5) + K += 5; - /* Move down to proper range now rounding is done */ - fll->k = K / 10; + /* Move down to proper range now rounding is done */ + fll->k = K / 10; - pr_debug("N=%x K=%x\n", fll->n, fll->k); + pr_debug("N=%x K=%x\n", fll->n, fll->k); + + default: + gcd_fll = gcd(freq_out, freq_in); + + fll->k = (freq_out - (freq_in * fll->n)) / gcd_fll; + fll->lambda = freq_in / gcd_fll; + + } return 0; } @@ -2139,9 +2151,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, * analysis bugs spewing warnings. */ if (freq_out) - ret = wm8994_get_fll_config(&fll, freq_in, freq_out); + ret = wm8994_get_fll_config(control, &fll, freq_in, freq_out); else - ret = wm8994_get_fll_config(&fll, wm8994->fll[id].in, + ret = wm8994_get_fll_config(control, &fll, wm8994->fll[id].in, wm8994->fll[id].out); if (ret < 0) return ret; @@ -2186,6 +2198,17 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_N_MASK, fll.n << WM8994_FLL1_N_SHIFT); + if (fll.lambda) { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_1 + reg_offset, + WM8958_FLL1_LAMBDA_MASK, + fll.lambda); + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, WM8958_FLL1_EFS_ENA); + } else { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, 0); + } + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | -- cgit v1.2.1 From 62477adf5f4ede918a97e648a5173b00bbbb17cc Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 13 May 2013 13:30:56 +0800 Subject: ASoC: mxs: move to use generic DMA helper With mxs-dma converted to generic DMA bindings, let's move mxs-pcm to use it by removing flages SND_DMAENGINE_PCM_FLAG_NO_DT and SND_DMAENGINE_PCM_FLAG_COMPAT. As the result, those mxs custom dma params code can be removed now. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 18 ------------------ sound/soc/mxs/mxs-pcm.h | 7 ------- sound/soc/mxs/mxs-saif.c | 29 +---------------------------- sound/soc/mxs/mxs-saif.h | 1 - 4 files changed, 1 insertion(+), 54 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b41fffc056fb..b16abbbf7764 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -49,24 +49,8 @@ static const struct snd_pcm_hardware snd_mxs_hardware = { .fifo_size = 32, }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_params *dma_params = param; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &dma_params->dma_data; - - return true; -} - static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { .pcm_hardware = &snd_mxs_hardware, - .compat_filter_fn = filter, .prealloc_buffer_size = 64 * 1024, }; @@ -74,8 +58,6 @@ int mxs_pcm_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_COMPAT | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 3aa918f9ed3e..bc685b67cac7 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,13 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -#include - -struct mxs_pcm_dma_params { - struct mxs_dma_data dma_data; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 71a972f5af97..49d870034bc3 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include #include @@ -604,8 +603,6 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); - dai->playback_dma_data = &saif->dma_param; - dai->capture_dma_data = &saif->dma_param; return 0; } @@ -664,7 +661,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; int ret = 0; struct device_node *master; @@ -719,22 +716,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -751,14 +732,6 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.dma_data.chan_irq < 0) { - ret = saif->dma_param.dma_data.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } - platform_set_drvdata(pdev, saif); ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e5bc90..53eaa4bf0e27 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -117,7 +117,6 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- cgit v1.2.1 From 785d81e29bd237b4e76ca27c3ebcc3281e663596 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:04 +0200 Subject: ASoC: ep93xx: Setup dma data in DAI probe This allows us to access the DAI DMA data when we create the PCM. We'll use this when converting ep39xx to generic DMA engine PCM driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 14 ++++---------- sound/soc/cirrus/ep93xx-i2s.c | 14 ++++---------- 2 files changed, 8 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 840c9b18201e..3f4f88877c84 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -314,22 +314,15 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, return 0; } -static int ep93xx_ac97_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int ep93xx_ac97_dai_probe(struct snd_soc_dai *dai) { - struct ep93xx_dma_data *dma_data; + dai->playback_dma_data = &ep93xx_ac97_pcm_out; + dai->capture_dma_data = &ep93xx_ac97_pcm_in; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &ep93xx_ac97_pcm_out; - else - dma_data = &ep93xx_ac97_pcm_in; - - snd_soc_dai_set_dma_data(dai, substream, dma_data); return 0; } static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { - .startup = ep93xx_ac97_startup, .trigger = ep93xx_ac97_trigger, }; @@ -337,6 +330,7 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = { .name = "ep93xx-ac97", .id = 0, .ac97_control = 1, + .probe = ep93xx_ac97_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 5c1102e9e159..ef731e687ebb 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -60,7 +60,6 @@ struct ep93xx_i2s_info { struct clk *mclk; struct clk *sclk; struct clk *lrclk; - struct ep93xx_dma_data *dma_data; void __iomem *regs; }; @@ -139,15 +138,11 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) } } -static int ep93xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + dai->playback_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; - snd_soc_dai_set_dma_data(cpu_dai, substream, - &info->dma_data[substream->stream]); return 0; } @@ -338,7 +333,6 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) #endif static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { - .startup = ep93xx_i2s_startup, .shutdown = ep93xx_i2s_shutdown, .hw_params = ep93xx_i2s_hw_params, .set_sysclk = ep93xx_i2s_set_sysclk, @@ -349,6 +343,7 @@ static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { static struct snd_soc_dai_driver ep93xx_i2s_dai = { .symmetric_rates= 1, + .probe = ep93xx_i2s_dai_probe, .suspend = ep93xx_i2s_suspend, .resume = ep93xx_i2s_resume, .playback = { @@ -407,7 +402,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, info); - info->dma_data = ep93xx_i2s_dma_data; err = snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component, &ep93xx_i2s_dai, 1); -- cgit v1.2.1 From e27e8a60cb4ca8e3b047c5d6ee9afff9c4c5b61a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:05 +0200 Subject: ASoC: ep93xx: Use generic dmaengine PCM Use the generic dmaengine PCM driver instead of a custom implementation. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-pcm.c | 138 ++++-------------------------------------- 2 files changed, 12 insertions(+), 128 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 88143db7e753..2c20f01e1f7e 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,7 +1,7 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" depends on ARCH_EP93XX && SND_SOC - select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the EP93xx I2S or AC97 interfaces. diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 488032690378..0e9f56e0d4b2 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -14,20 +14,14 @@ #include #include -#include -#include +#include #include -#include -#include #include -#include #include #include #include -#include -#include static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | @@ -63,134 +57,24 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) return false; } -static int ep93xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); - - return snd_dmaengine_pcm_open_request_chan(substream, - ep93xx_pcm_dma_filter, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); -} - -static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - return 0; -} - -static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops ep93xx_pcm_ops = { - .open = ep93xx_pcm_open, - .close = snd_dmaengine_pcm_close_release_chan, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = ep93xx_pcm_hw_params, - .hw_free = ep93xx_pcm_hw_free, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = ep93xx_pcm_mmap, -}; - -static int ep93xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ep93xx_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - buf->bytes = size; - - return (buf->area == NULL) ? -ENOMEM : 0; -} - -static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, - buf->addr); - buf->area = NULL; - } -} - -static u64 ep93xx_pcm_dmamask = DMA_BIT_MASK(32); - -static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &ep93xx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = ep93xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = ep93xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver ep93xx_soc_platform = { - .ops = &ep93xx_pcm_ops, - .pcm_new = &ep93xx_pcm_new, - .pcm_free = &ep93xx_pcm_free_dma_buffers, +static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { + .pcm_hardware = &ep93xx_pcm_hardware, + .compat_filter_fn = ep93xx_pcm_dma_filter, + .prealloc_buffer_size = 131072, }; static int ep93xx_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &ep93xx_soc_platform); + return snd_dmaengine_pcm_register(&pdev->dev, + &ep93xx_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT); } static int ep93xx_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_platform(&pdev->dev); + snd_dmaengine_pcm_unregister(&pdev->dev); return 0; } -- cgit v1.2.1 From 110147c8c5136e1768a382da8896cf7f8b518982 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 13 May 2013 13:26:12 -0600 Subject: ASoC: tegra: always use clk_get() in utility code Now that all of the Tegra device trees have been updated to represent the required audio clocks, remove the compatibility code from the Tegra ASoC utility code, and always use clk_get() rather than clk_get_sys(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 23 +++++------------------ 1 file changed, 5 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 24fb001be7f4..d173880f290d 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -173,7 +173,6 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) { int ret; - bool new_clocks = false; data->dev = dev; @@ -181,40 +180,28 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (of_machine_is_compatible("nvidia,tegra114")) { + else if (of_machine_is_compatible("nvidia,tegra114")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; - new_clocks = true; - } else { + else { dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; } - if (new_clocks) - data->clk_pll_a = clk_get(dev, "pll_a"); - else - data->clk_pll_a = clk_get_sys(NULL, "pll_a"); + data->clk_pll_a = clk_get(dev, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); ret = PTR_ERR(data->clk_pll_a); goto err; } - if (new_clocks) - data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); - else - data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); + data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); if (IS_ERR(data->clk_pll_a_out0)) { dev_err(data->dev, "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(data->clk_pll_a_out0); goto err_put_pll_a; } - if (new_clocks) - data->clk_cdev1 = clk_get(dev, "mclk"); - else if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); - else - data->clk_cdev1 = clk_get_sys("extern1", NULL); + data->clk_cdev1 = clk_get(dev, "mclk"); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); -- cgit v1.2.1 From 9dbce04402e33e362e3e946c437bc70b8102a95d Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 14 May 2013 15:02:44 +0300 Subject: ASoC: wm_adsp: memory leak in wm_adsp_create_control() There are two return paths which don't kfree(name). Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 137830611928..d715c8ede772 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -836,7 +836,8 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, region_name = "ZM"; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_name; } snprintf(name, PAGE_SIZE, "DSP%d %s %x", @@ -847,7 +848,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, if (!strcmp(ctl->name, name)) { if (!ctl->enabled) ctl->enabled = 1; - return 0; + goto found; } } @@ -887,6 +888,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); schedule_work(&ctl_work->work); +found: kfree(name); return 0; -- cgit v1.2.1 From 90996f43b3fa36075e501a52a3e2286896d74e79 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 11:05:30 +0200 Subject: ASoC: core: Move snd_soc_set_runtime_hwparams() to soc-pcm.c snd_soc_set_runtime_hwparams() is the only PCM related function that lives in soc-core.c. All other PCM related functions live in soc-pcm.c, so move snd_soc_set_runtime_hwparams() over as well for a bit more consistency. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 23 ----------------------- sound/soc/soc-pcm.c | 23 +++++++++++++++++++++++ 2 files changed, 23 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 308895a438d6..9d95ef531308 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2219,29 +2219,6 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, } EXPORT_SYMBOL_GPL(snd_soc_test_bits); -/** - * snd_soc_set_runtime_hwparams - set the runtime hardware parameters - * @substream: the pcm substream - * @hw: the hardware parameters - * - * Sets the substream runtime hardware parameters. - */ -int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, - const struct snd_pcm_hardware *hw) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.info = hw->info; - runtime->hw.formats = hw->formats; - runtime->hw.period_bytes_min = hw->period_bytes_min; - runtime->hw.period_bytes_max = hw->period_bytes_max; - runtime->hw.periods_min = hw->periods_min; - runtime->hw.periods_max = hw->periods_max; - runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; - runtime->hw.fifo_size = hw->fifo_size; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); - /** * snd_soc_cnew - create new control * @_template: control template diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8eefa491..058b40378451 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -33,6 +33,29 @@ #define DPCM_MAX_BE_USERS 8 +/** + * snd_soc_set_runtime_hwparams - set the runtime hardware parameters + * @substream: the pcm substream + * @hw: the hardware parameters + * + * Sets the substream runtime hardware parameters. + */ +int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, + const struct snd_pcm_hardware *hw) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + runtime->hw.info = hw->info; + runtime->hw.formats = hw->formats; + runtime->hw.period_bytes_min = hw->period_bytes_min; + runtime->hw.period_bytes_max = hw->period_bytes_max; + runtime->hw.periods_min = hw->periods_min; + runtime->hw.periods_max = hw->periods_max; + runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; + runtime->hw.fifo_size = hw->fifo_size; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); + /* DPCM stream event, send event to FE and all active BEs. */ static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) -- cgit v1.2.1 From bd477c31ca3ae85645fb2852bfa3954a623f9237 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 11:05:31 +0200 Subject: ASoC: core: Add helper function to initialize the runtime pcm We use the same code to initialize the runtime pcm based on the snd_soc_pcm_stream struct on both the playback and capture path. Factor this code into a helper function to make things a bit more tidy. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 68 +++++++++++++++++++---------------------------------- 1 file changed, 24 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 058b40378451..2f6f545f4ee8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -147,6 +147,26 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } +static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, + struct snd_soc_pcm_stream *codec_stream, + struct snd_soc_pcm_stream *cpu_stream) +{ + hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min); + hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max); + hw->channels_min = max(codec_stream->channels_min, + cpu_stream->channels_min); + hw->channels_max = min(codec_stream->channels_max, + cpu_stream->channels_max); + hw->formats = codec_stream->formats & cpu_stream->formats; + hw->rates = codec_stream->rates & cpu_stream->rates; + if (codec_stream->rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + hw->rates |= cpu_stream->rates; + if (cpu_stream->rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + hw->rates |= codec_stream->rates; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -212,51 +232,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = - max(codec_dai_drv->playback.rate_min, - cpu_dai_drv->playback.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->playback.rate_max, - cpu_dai_drv->playback.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->playback.channels_min, - cpu_dai_drv->playback.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->playback.channels_max, - cpu_dai_drv->playback.channels_max); - runtime->hw.formats = - codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; - runtime->hw.rates = - codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; - if (codec_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->playback.rates; - if (cpu_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->playback.rates; + soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback, + &cpu_dai_drv->playback); } else { - runtime->hw.rate_min = - max(codec_dai_drv->capture.rate_min, - cpu_dai_drv->capture.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->capture.rate_max, - cpu_dai_drv->capture.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->capture.channels_min, - cpu_dai_drv->capture.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->capture.channels_max, - cpu_dai_drv->capture.channels_max); - runtime->hw.formats = - codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; - runtime->hw.rates = - codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; - if (codec_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->capture.rates; - if (cpu_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->capture.rates; + soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture, + &cpu_dai_drv->capture); } ret = -EINVAL; -- cgit v1.2.1 From 2b581074357c42f63ae827ee28c9f244b91a38ac Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 11:05:32 +0200 Subject: ASoC: core: Use kasprintf instead of opencoding it kasprintf calculates the size of the result string, allocates a buffer large enough to hold the string and then performs the format string operation. There are a couple of places in ASoC where these three steps are done by hand and where kasprintf can be used instead. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 +----- sound/soc/soc-dapm.c | 31 ++++++++++--------------------- 2 files changed, 11 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9d95ef531308..4489c5b7b53a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2237,7 +2237,6 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, struct snd_kcontrol_new template; struct snd_kcontrol *kcontrol; char *name = NULL; - int name_len; memcpy(&template, _template, sizeof(template)); template.index = 0; @@ -2246,13 +2245,10 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, long_name = template.name; if (prefix) { - name_len = strlen(long_name) + strlen(prefix) + 2; - name = kmalloc(name_len, GFP_KERNEL); + name = kasprintf(GFP_KERNEL, "%s %s", prefix, long_name); if (!name) return NULL; - snprintf(name, name_len, "%s %s", prefix, long_name); - template.name = name; } else { template.name = long_name; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e4e5420de725..071579be7cb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -521,7 +521,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, int wlistentries; size_t wlistsize; bool wname_in_long_name, kcname_in_long_name; - size_t name_len; char *long_name; const char *name; int ret; @@ -586,25 +585,19 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, } if (wname_in_long_name && kcname_in_long_name) { - name_len = strlen(w->name) - prefix_len + 1 + - strlen(w->kcontrol_news[kci].name) + 1; - - long_name = kmalloc(name_len, GFP_KERNEL); - if (long_name == NULL) { - kfree(wlist); - return -ENOMEM; - } - /* * The control will get a prefix from the control * creation process but we're also using the same * prefix for widgets so cut the prefix off the * front of the widget name. */ - snprintf(long_name, name_len, "%s %s", + long_name = kasprintf(GFP_KERNEL, "%s %s", w->name + prefix_len, w->kcontrol_news[kci].name); - long_name[name_len - 1] = '\0'; + if (long_name == NULL) { + kfree(wlist); + return -ENOMEM; + } name = long_name; } else if (wname_in_long_name) { @@ -3077,7 +3070,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; - size_t name_len; int ret; if ((w = dapm_cnew_widget(widget)) == NULL) @@ -3118,19 +3110,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; } - name_len = strlen(widget->name) + 1; if (dapm->codec && dapm->codec->name_prefix) - name_len += 1 + strlen(dapm->codec->name_prefix); - w->name = kmalloc(name_len, GFP_KERNEL); + w->name = kasprintf(GFP_KERNEL, "%s %s", + dapm->codec->name_prefix, widget->name); + else + w->name = kasprintf(GFP_KERNEL, "%s", widget->name); + if (w->name == NULL) { kfree(w); return NULL; } - if (dapm->codec && dapm->codec->name_prefix) - snprintf((char *)w->name, name_len, "%s %s", - dapm->codec->name_prefix, widget->name); - else - snprintf((char *)w->name, name_len, "%s", widget->name); switch (w->id) { case snd_soc_dapm_switch: -- cgit v1.2.1 From be87f75efed8248d74c8ec56e997de989ecc963e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:47 +0200 Subject: ASoC: ep93xx-i2s: Staticize non exported struct The ep93xx_i2s_dma_data struct is not used outside of ep93xx-i2s.c, so make it static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index ef731e687ebb..17ad70bca9fe 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -63,7 +63,7 @@ struct ep93xx_i2s_info { void __iomem *regs; }; -struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { +static struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { [SNDRV_PCM_STREAM_PLAYBACK] = { .name = "i2s-pcm-out", .port = EP93XX_DMA_I2S1, -- cgit v1.2.1 From 5d0c8a58747c84efe93fb632ae25eb377aea1fa0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:48 +0200 Subject: ASoC: kirkwood-dma: Staticize non exported struct The kirkwood_dma_ops struct is not used outside of kirkwood-dma.c, so make it static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index d3d4bdca1cc6..a9f14530c3db 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -289,7 +289,7 @@ static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream return count; } -struct snd_pcm_ops kirkwood_dma_ops = { +static struct snd_pcm_ops kirkwood_dma_ops = { .open = kirkwood_dma_open, .close = kirkwood_dma_close, .ioctl = snd_pcm_lib_ioctl, -- cgit v1.2.1 From 169cc48982f2583de1fea89e7becb1304730a34e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:49 +0200 Subject: ASoC: spear: spdif_in: Staticize non exported struct The spdif_in_dai struct is not used outside of spdif_in.c, so make it static. Signed-off-by: Lars-Peter Clausen Acked-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/spear/spdif_in.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 643ada6a1fa5..f0071ddbfa7d 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -152,7 +152,7 @@ static struct snd_soc_dai_ops spdif_in_dai_ops = { .hw_params = spdif_in_hw_params, }; -struct snd_soc_dai_driver spdif_in_dai = { +static struct snd_soc_dai_driver spdif_in_dai = { .probe = spdif_in_dai_probe, .capture = { .channels_min = 2, -- cgit v1.2.1 From 19c7efcd24a71d0f707cecd3c0cee5f4a1289b1e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:50 +0200 Subject: ASoC: spear: spdif_out: Staticize unexported function The spdif_soc_dai_probe function is only used in spdif_out.c, so make it static. Signed-off-by: Lars-Peter Clausen Acked-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/spear/spdif_out.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index fd25cc5efe27..4bde5123cea6 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -240,7 +240,7 @@ static const struct snd_kcontrol_new spdif_out_controls[] = { spdif_mute_get, spdif_mute_put), }; -int spdif_soc_dai_probe(struct snd_soc_dai *dai) +static int spdif_soc_dai_probe(struct snd_soc_dai *dai) { struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); -- cgit v1.2.1 From 60e10d2fb0f0739a5862311258e10520accc9259 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:51 +0200 Subject: ASoC: mmp-pcm: Staticize non exported structs and functions The mmp_pcm_ops and mmp_soc_platform struct as well as the mmp_pcm_new() function are only used in mmp-pcm.c, so make them static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 349930015264..5d57e071cdf5 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -147,7 +147,7 @@ static int mmp_pcm_mmap(struct snd_pcm_substream *substream, vma->vm_end - vma->vm_start, vma->vm_page_prot); } -struct snd_pcm_ops mmp_pcm_ops = { +static struct snd_pcm_ops mmp_pcm_ops = { .open = mmp_pcm_open, .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, @@ -208,7 +208,7 @@ static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, return 0; } -int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) +static int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm_substream *substream; struct snd_pcm *pcm = rtd->pcm; @@ -229,7 +229,7 @@ err: return ret; } -struct snd_soc_platform_driver mmp_soc_platform = { +static struct snd_soc_platform_driver mmp_soc_platform = { .ops = &mmp_pcm_ops, .pcm_new = mmp_pcm_new, .pcm_free = mmp_pcm_free_dma_buffers, -- cgit v1.2.1 From 5d9ff402152fe9421c5ed86b4f651a8c62de7c7a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:52 +0200 Subject: ASoC: mmp-sspa: Staticize non exported struct The mmp_sspa_dai struct is only used in mmp-sspa.c, so make it static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-sspa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index a64779980177..62142ce367c7 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -388,7 +388,7 @@ static struct snd_soc_dai_ops mmp_sspa_dai_ops = { .set_fmt = mmp_sspa_set_dai_fmt, }; -struct snd_soc_dai_driver mmp_sspa_dai = { +static struct snd_soc_dai_driver mmp_sspa_dai = { .probe = mmp_sspa_probe, .playback = { .channels_min = 1, -- cgit v1.2.1 From 19b6317b8c78a51c41b3d7abf14fdbb0df8f41b7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:53 +0200 Subject: ASoC: bf5xx-tdm-pcm: Staticize non exported struct The bf5xx_tdm_pcm_ops struct is only used in bf5xx-tdm-pcm.c, so make it static. Cc: Scott Jiang Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-tdm-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 0e6b888bb4cc..19e855d5677e 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -229,8 +229,7 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, return 0; } - -struct snd_pcm_ops bf5xx_pcm_tdm_ops = { +static struct snd_pcm_ops bf5xx_pcm_tdm_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, -- cgit v1.2.1 From cdeecac4e610ce1094497740ef84aa837e2e874f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:54 +0200 Subject: ASoC: mop500_ab8500: Staticize non exported functions The mop500_ab8500_startup(), the mop500_ab8500_shutdown() and the mop500_ab8500_hw_params() function are not used outside of mop500_ab8500, so make them static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 44e25a291044..884a36224fb1 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -183,7 +183,7 @@ static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { /* ASoC */ -int mop500_ab8500_startup(struct snd_pcm_substream *substream) +static int mop500_ab8500_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -192,7 +192,7 @@ int mop500_ab8500_startup(struct snd_pcm_substream *substream) snd_soc_card_get_drvdata(rtd->card)); } -void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +static void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct device *dev = rtd->card->dev; @@ -206,7 +206,7 @@ void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) rx_slots = DEF_RX_SLOTS; } -int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, +static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; -- cgit v1.2.1 From aba1e2be4dcfda7069cc0b82c73b89707595a454 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:55 +0200 Subject: ASoC: mop500: Staticize non exported struct The mop500_dai_links struct is not used outside of mop500.c, so make it static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/ux500/mop500.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 204b899c2311..8f5cd00a6e46 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -27,7 +27,7 @@ #include "mop500_ab8500.h" /* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ -struct snd_soc_dai_link mop500_dai_links[] = { +static struct snd_soc_dai_link mop500_dai_links[] = { { .name = "ab8500_0", .stream_name = "ab8500_0", -- cgit v1.2.1 From 3eb28d3ca8f0c94ae40f57fbd53ef3805c8fdd2d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:56 +0200 Subject: ASoC: sn95031: Staticize non exported struct The sn95031_codec struct is not used outside of sn95031.c, so make it static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index d1ae869d3181..dba26e63844e 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -883,7 +883,7 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sn95031_codec = { +static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, .read = sn95031_read, -- cgit v1.2.1 From cde11aedea7325d9e48c9f6f7ac468287d457e79 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 14 May 2013 22:19:57 +0200 Subject: ASoC: davinci-sffsdr: Staticize non exported struct The pcm3008_codec struct is not used outside of davinci-sffsdr.c, so make it static. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-sffsdr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 5be65aae7e0e..24df5bcf3bbc 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -106,7 +106,7 @@ static struct pcm3008_setup_data sffsdr_pcm3008_setup = { .pdda_pin = GPIO(38), }; -struct platform_device pcm3008_codec = { +static struct platform_device pcm3008_codec = { .name = "pcm3008-codec", .id = 0, .dev = { -- cgit v1.2.1 From 7ae6871fe51e337caa88025ac2dc0c586c4d4a09 Mon Sep 17 00:00:00 2001 From: Paul Bolle Date: Tue, 14 May 2013 17:07:21 +0200 Subject: ASoC: remove saarb and tavorevb3 machine drivers Support for PXA95x was removed in v3.8. This means that the Kconfig symbols MACH_SAARB and MACH_TAVOREVB3 are no longer available. This leaves the SoC Audio support for Marvell Saarb and Marvell Tavor EVB3 unbuildable. Remove these drivers too. Signed-off-by: Paul Bolle Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 20 ----- sound/soc/pxa/Makefile | 4 - sound/soc/pxa/saarb.c | 190 ---------------------------------------------- sound/soc/pxa/tavorevb3.c | 189 --------------------------------------------- 4 files changed, 403 deletions(-) delete mode 100644 sound/soc/pxa/saarb.c delete mode 100644 sound/soc/pxa/tavorevb3.c (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4d2e46fae77c..b35809467547 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -130,26 +130,6 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. -config SND_SOC_SAARB - tristate "SoC Audio support for Marvell Saarb" - depends on SND_PXA2XX_SOC && MACH_SAARB - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - -config SND_SOC_TAVOREVB3 - tristate "SoC Audio support for Marvell Tavor EVB3" - depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - config SND_PXA910_SOC tristate "SoC Audio for Marvell PXA910 chip" depends on ARCH_MMP && SND diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index d8a265d2d5d7..2cff67b61dc3 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,8 +23,6 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o -snd-soc-saarb-objs := saarb.o -snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-hx4700-objs := hx4700.o snd-soc-magician-objs := magician.o @@ -48,8 +46,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o -obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o -obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c deleted file mode 100644 index c34146b776b4..000000000000 --- a/sound/soc/pxa/saarb.c +++ /dev/null @@ -1,190 +0,0 @@ -/* - * saarb.c -- SoC audio for saarb - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *saarb_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* saarb machine dapm widgets */ -static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* saarb machine audio map */ -static const struct snd_soc_dapm_route saarb_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - - return ret; -} - -static struct snd_soc_ops saarb_i2s_ops = { - .hw_params = saarb_i2s_hw_params, -}; - -static struct snd_soc_dai_link saarb_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = saarb_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &saarb_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_saarb = { - .name = "Saarb", - .owner = THIS_MODULE, - .dai_link = saarb_dai, - .num_links = ARRAY_SIZE(saarb_dai), - - .dapm_widgets = saarb_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), - .dapm_routes = saarb_audio_map, - .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), -}; - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init saarb_init(void) -{ - int ret; - - if (!machine_is_saarb()) - return -ENODEV; - saarb_snd_device = platform_device_alloc("soc-audio", -1); - if (!saarb_snd_device) - return -ENOMEM; - - platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); - - ret = platform_device_add(saarb_snd_device); - if (ret) - platform_device_put(saarb_snd_device); - - return ret; -} - -static void __exit saarb_exit(void) -{ - platform_device_unregister(saarb_snd_device); -} - -module_init(saarb_init); -module_exit(saarb_exit); - -MODULE_AUTHOR("Haojian Zhuang "); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c deleted file mode 100644 index 8b5ab8f72726..000000000000 --- a/sound/soc/pxa/tavorevb3.c +++ /dev/null @@ -1,189 +0,0 @@ -/* - * tavorevb3.c -- SoC audio for Tavor EVB3 - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *evb3_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* tavorevb3 machine dapm widgets */ -static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic 2", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* tavorevb3 machine audio map */ -static const struct snd_soc_dapm_route evb3_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int evb3_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - return ret; -} - -static struct snd_soc_ops evb3_i2s_ops = { - .hw_params = evb3_i2s_hw_params, -}; - -static struct snd_soc_dai_link evb3_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = evb3_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &evb3_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_evb3 = { - .name = "Tavor EVB3", - .owner = THIS_MODULE, - .dai_link = evb3_dai, - .num_links = ARRAY_SIZE(evb3_dai), - - .dapm_widgets = evb3_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets), - .dapm_routes = evb3_audio_map, - .num_dapm_routes = ARRAY_SIZE(evb3_audio_map), -}; - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init tavorevb3_init(void) -{ - int ret; - - if (!machine_is_tavorevb3()) - return -ENODEV; - evb3_snd_device = platform_device_alloc("soc-audio", -1); - if (!evb3_snd_device) - return -ENOMEM; - - platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); - - ret = platform_device_add(evb3_snd_device); - if (ret) - platform_device_put(evb3_snd_device); - - return ret; -} - -static void __exit tavorevb3_exit(void) -{ - platform_device_unregister(evb3_snd_device); -} - -module_init(tavorevb3_init); -module_exit(tavorevb3_exit); - -MODULE_AUTHOR("Haojian Zhuang "); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); -MODULE_LICENSE("GPL"); -- cgit v1.2.1 From bd41bc9696b5631b2c2fe26f40c8cdd99b3aeb3e Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 25 Apr 2013 11:18:46 +0800 Subject: ASoC: fsl: remove use of imx-pcm-audio from fsl_ssi Rather than instantiating imx-pcm-audio to call imx_pcm_dma_init(), fsl_ssi can just directly call it to save the use of imx-pcm-audio. With this change, fsl_ssi becomes not only a cpu DAI but also a platform device, so updates platform device setup in imx-sgtl5000 accordingly. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 13 ++++--------- sound/soc/fsl/imx-sgtl5000.c | 2 +- 2 files changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0f0bed6def9e..2f2d837df07f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -122,7 +122,6 @@ struct fsl_ssi_private { bool new_binding; bool ssi_on_imx; struct clk *clk; - struct platform_device *imx_pcm_pdev; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; struct imx_dma_data filter_data_tx; @@ -809,13 +808,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (ssi_private->ssi_on_imx) { - ssi_private->imx_pcm_pdev = - platform_device_register_simple("imx-pcm-audio", - -1, NULL, 0); - if (IS_ERR(ssi_private->imx_pcm_pdev)) { - ret = PTR_ERR(ssi_private->imx_pcm_pdev); + ret = imx_pcm_dma_init(pdev); + if (ret) goto error_dev; - } } /* @@ -854,7 +849,7 @@ done: error_dai: if (ssi_private->ssi_on_imx) - platform_device_unregister(ssi_private->imx_pcm_pdev); + imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); error_dev: @@ -889,7 +884,7 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); if (ssi_private->ssi_on_imx) { - platform_device_unregister(ssi_private->imx_pcm_pdev); + imx_pcm_dma_exit(pdev); clk_disable_unprepare(ssi_private->clk); clk_put(ssi_private->clk); } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 5a6aaa3b947a..a60aaa053d28 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -149,7 +149,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->dai.codec_dai_name = "sgtl5000"; data->dai.codec_of_node = codec_np; data->dai.cpu_of_node = ssi_np; - data->dai.platform_name = "imx-pcm-audio"; + data->dai.platform_of_node = ssi_np; data->dai.init = &imx_sgtl5000_dai_init; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; -- cgit v1.2.1 From 3b7d46380beae3de4a0f03ba4dcbd509c97ab503 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 25 Apr 2013 11:18:47 +0800 Subject: ASoC: fsl: remove use of imx-pcm-audio from imx-ssi Rather than instantiating imx-pcm-audio to call imx_pcm_dma_init(), imx-ssi can just directly call it to save the use of imx-pcm-audio. With this change, imx-ssi becomes not only a cpu DAI but also a platform device, so updates platform device setup in imx-mc13783 and mx27vis-aic32x4 accordingly. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-ssi.c | 21 +++++---------------- sound/soc/fsl/imx-ssi.h | 1 - sound/soc/fsl/mx27vis-aic32x4.c | 2 +- 4 files changed, 7 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 4ae30f21fdb5..9df173c091a6 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -64,7 +64,7 @@ static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = { .codec_dai_name = "mc13783-hifi", .codec_name = "mc13783-codec", .cpu_dai_name = "imx-ssi.0", - .platform_name = "imx-pcm-audio.0", + .platform_name = "imx-ssi.0", .ops = &imx_mc13783_hifi_ops, .symmetric_rates = 1, .dai_fmt = FMT_SSI, diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 902fab02b851..b5a2b040816c 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -608,24 +608,13 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_pdev_fiq_add; } - ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev) { - ret = -ENOMEM; - goto failed_pdev_alloc; - } - - platform_set_drvdata(ssi->soc_platform_pdev, ssi); - ret = platform_device_add(ssi->soc_platform_pdev); - if (ret) { - dev_err(&pdev->dev, "failed to add platform device\n"); - goto failed_pdev_add; - } + ret = imx_pcm_dma_init(pdev); + if (ret) + goto failed_pcm_dma; return 0; -failed_pdev_add: - platform_device_put(ssi->soc_platform_pdev); -failed_pdev_alloc: +failed_pcm_dma: platform_device_del(ssi->soc_platform_pdev_fiq); failed_pdev_fiq_add: platform_device_put(ssi->soc_platform_pdev_fiq); @@ -645,7 +634,7 @@ static int imx_ssi_remove(struct platform_device *pdev) struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); - platform_device_unregister(ssi->soc_platform_pdev); + imx_pcm_dma_exit(pdev); platform_device_unregister(ssi->soc_platform_pdev_fiq); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index bb6b3dbb13fd..b052fad8f6c7 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -212,7 +212,6 @@ struct imx_ssi { int enabled; - struct platform_device *soc_platform_pdev; struct platform_device *soc_platform_pdev_fiq; }; diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index 3d1074179057..f4c3bda5e69e 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -161,7 +161,7 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .name = "tlv320aic32x4", .stream_name = "TLV320AIC32X4", .codec_dai_name = "tlv320aic32x4-hifi", - .platform_name = "imx-pcm-audio.0", + .platform_name = "imx-ssi.0", .codec_name = "tlv320aic32x4.0-0018", .cpu_dai_name = "imx-ssi.0", .ops = &mx27vis_aic32x4_snd_ops, -- cgit v1.2.1 From 88e89f5548a6e19bf837633f622764f2d1531748 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 25 Apr 2013 11:18:48 +0800 Subject: ASoC: fsl: create function imx_pcm_fiq_exit() Create function imx_pcm_fiq_exit() to be paired with imx_pcm_fiq_init() just like the pair of imx_pcm_dma_init() and imx_pcm_dma_exit(). Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-fiq.c | 5 +++++ sound/soc/fsl/imx-pcm.c | 2 +- sound/soc/fsl/imx-pcm.h | 5 +++++ 3 files changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 670b96b0ce2f..710c06990450 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -314,3 +314,8 @@ failed_register: return ret; } + +void imx_pcm_fiq_exit(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); +} diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index c49896442d8e..16a956bcc52b 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -115,7 +115,7 @@ static int imx_pcm_probe(struct platform_device *pdev) static int imx_pcm_remove(struct platform_device *pdev) { if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0) - snd_soc_unregister_platform(&pdev->dev); + imx_pcm_fiq_exit(pdev); else imx_pcm_dma_exit(pdev); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index b7fa0d75c687..073bf389c02e 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -53,11 +53,16 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) #ifdef CONFIG_SND_SOC_IMX_PCM_FIQ int imx_pcm_fiq_init(struct platform_device *pdev); +void imx_pcm_fiq_exit(struct platform_device *pdev); #else static inline int imx_pcm_fiq_init(struct platform_device *pdev) { return -ENODEV; } + +static inline void imx_pcm_fiq_exit(struct platform_device *pdev) +{ +} #endif #endif /* _IMX_PCM_H */ -- cgit v1.2.1 From 2bf9d4bbd0fa97ff6f214484f62fc8aca64d1d00 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 25 Apr 2013 11:18:49 +0800 Subject: ASoC: fsl: remove use of imx-fiq-pcm-audio from imx-ssi Rather than instantiating imx-fiq-pcm-audio to call imx_pcm_fiq_init(), imx-ssi can just directly call it to save the use of imx-fiq-pcm-audio. With this change, imx-ssi becomes not only a cpu DAI but also a platform device, so updates platform device setup in eukrea-tlv320, phycore-ac97 and wm1133-ev1 accordingly. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/imx-ssi.c | 23 ++++++----------------- sound/soc/fsl/imx-ssi.h | 2 -- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- 5 files changed, 9 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 75ffdf0e2aad..9a4a0ca2c1de 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -80,7 +80,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-fiq-pcm-audio.0", + .platform_name = "imx-ssi.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index b5a2b040816c..1b2e750151ae 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -595,18 +595,9 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_register; } - ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev_fiq) { - ret = -ENOMEM; - goto failed_pdev_fiq_alloc; - } - - platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi); - ret = platform_device_add(ssi->soc_platform_pdev_fiq); - if (ret) { - dev_err(&pdev->dev, "failed to add platform device\n"); - goto failed_pdev_fiq_add; - } + ret = imx_pcm_fiq_init(pdev); + if (ret) + goto failed_pcm_fiq; ret = imx_pcm_dma_init(pdev); if (ret) @@ -615,10 +606,8 @@ static int imx_ssi_probe(struct platform_device *pdev) return 0; failed_pcm_dma: - platform_device_del(ssi->soc_platform_pdev_fiq); -failed_pdev_fiq_add: - platform_device_put(ssi->soc_platform_pdev_fiq); -failed_pdev_fiq_alloc: + imx_pcm_fiq_exit(pdev); +failed_pcm_fiq: snd_soc_unregister_component(&pdev->dev); failed_register: release_mem_region(res->start, resource_size(res)); @@ -635,7 +624,7 @@ static int imx_ssi_remove(struct platform_device *pdev) struct imx_ssi *ssi = platform_get_drvdata(pdev); imx_pcm_dma_exit(pdev); - platform_device_unregister(ssi->soc_platform_pdev_fiq); + imx_pcm_fiq_exit(pdev); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index b052fad8f6c7..d5003cefca8d 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -211,8 +211,6 @@ struct imx_ssi { struct imx_dma_data filter_data_rx; int enabled; - - struct platform_device *soc_platform_pdev_fiq; }; #endif /* _IMX_SSI_H */ diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index f8da6dd115ed..ae403c29688f 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -33,7 +33,7 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .cpu_dai_name = "imx-ssi.0", - .platform_name = "imx-fiq-pcm-audio.0", + .platform_name = "imx-ssi.0", .ops = &imx_phycore_hifi_ops, }, }; diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index fe54a69073e5..fce63252bdbb 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -245,7 +245,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { .stream_name = "Audio", .cpu_dai_name = "imx-ssi.0", .codec_dai_name = "wm8350-hifi", - .platform_name = "imx-fiq-pcm-audio.0", + .platform_name = "imx-ssi.0", .codec_name = "wm8350-codec.0-0x1a", .init = wm1133_ev1_init, .ops = &wm1133_ev1_ops, -- cgit v1.2.1 From dbdf6b54340e1671439a4a5efbd15b7a0b14eacb Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 25 Apr 2013 11:18:50 +0800 Subject: ASoC: fsl: remove imx-pcm driver With imx-pcm-dma moving to generic dmaengine pcm driver and the removal of imx-pcm-audio/imx-fiq-pcm-audio platform device use, now imx-pcm driver contains a few functions that are only used by imx-pcm-fiq.c. Move these functions into imx-pcm-fiq.c and remove imx-pcm.c completely. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 5 -- sound/soc/fsl/Makefile | 11 +--- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 87 ++++++++++++++++++++++++++ sound/soc/fsl/imx-pcm.c | 145 -------------------------------------------- sound/soc/fsl/imx-pcm.h | 5 -- 6 files changed, 91 insertions(+), 164 deletions(-) delete mode 100644 sound/soc/fsl/imx-pcm.c (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3843a18d4e56..7860cc27e5b2 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,18 +108,13 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM - tristate - config SND_SOC_IMX_PCM_FIQ bool select FIQ - select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA bool select SND_SOC_GENERIC_DMAENGINE_PCM - select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index afd34794db53..91883f8a2321 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -30,18 +30,11 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o -snd-soc-imx-pcm-objs := imx-pcm.o -ifneq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),) - snd-soc-imx-pcm-objs += imx-pcm-fiq.o -endif -ifneq ($(CONFIG_SND_SOC_IMX_PCM_DMA),) - snd-soc-imx-pcm-objs += imx-pcm-dma.o -endif - obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index c246fb514930..fde4d2ea68c8 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -67,8 +67,10 @@ int imx_pcm_dma_init(struct platform_device *pdev) SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } +EXPORT_SYMBOL_GPL(imx_pcm_dma_init); void imx_pcm_dma_exit(struct platform_device *pdev) { snd_dmaengine_pcm_unregister(&pdev->dev); } +EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 710c06990450..310d90290320 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -225,6 +225,22 @@ static int snd_imx_close(struct snd_pcm_substream *substream) return 0; } +static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} + static struct snd_pcm_ops imx_pcm_ops = { .open = snd_imx_open, .close = snd_imx_close, @@ -236,6 +252,54 @@ static struct snd_pcm_ops imx_pcm_ops = { .mmap = snd_imx_pcm_mmap, }; +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) @@ -268,6 +332,27 @@ static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) return 0; } +static void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + static void imx_pcm_fiq_free(struct snd_pcm *pcm) { mxc_set_irq_fiq(ssi_irq, 0); @@ -314,8 +399,10 @@ failed_register: return ret; } +EXPORT_SYMBOL_GPL(imx_pcm_fiq_init); void imx_pcm_fiq_exit(struct platform_device *pdev) { snd_soc_unregister_platform(&pdev->dev); } +EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit); diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c deleted file mode 100644 index 16a956bcc52b..000000000000 --- a/sound/soc/fsl/imx-pcm.c +++ /dev/null @@ -1,145 +0,0 @@ -/* - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include -#include -#include "imx-pcm.h" - -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - - pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); - return ret; -} -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); - -static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = IMX_SSI_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - - return 0; -} - -static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); - -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &imx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - -out: - return ret; -} -EXPORT_SYMBOL_GPL(imx_pcm_new); - -void imx_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(imx_pcm_free); - -static int imx_pcm_probe(struct platform_device *pdev) -{ - if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0) - return imx_pcm_fiq_init(pdev); - - return imx_pcm_dma_init(pdev); -} - -static int imx_pcm_remove(struct platform_device *pdev) -{ - if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0) - imx_pcm_fiq_exit(pdev); - else - imx_pcm_dma_exit(pdev); - - return 0; -} - -static struct platform_device_id imx_pcm_devtype[] = { - { .name = "imx-pcm-audio", }, - { .name = "imx-fiq-pcm-audio", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(platform, imx_pcm_devtype); - -static struct platform_driver imx_pcm_driver = { - .driver = { - .name = "imx-pcm", - .owner = THIS_MODULE, - }, - .id_table = imx_pcm_devtype, - .probe = imx_pcm_probe, - .remove = imx_pcm_remove, -}; -module_platform_driver(imx_pcm_driver); - -MODULE_DESCRIPTION("Freescale i.MX PCM driver"); -MODULE_AUTHOR("Sascha Hauer "); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 073bf389c02e..67f656c7c320 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -32,11 +32,6 @@ imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, dma_data->peripheral_type = IMX_DMATYPE_SSI; } -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma); -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); -void imx_pcm_free(struct snd_pcm *pcm); - #ifdef CONFIG_SND_SOC_IMX_PCM_DMA int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); -- cgit v1.2.1 From a64cbb949a18a8eefc40881e6e68734ca7275d36 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 16 May 2013 14:30:28 +0100 Subject: ASoC: wm5102: Correct OSR control name for EPOUT Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d3939eef..6e1ee130705a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -814,7 +814,7 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), -SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), +SOC_VALUE_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- cgit v1.2.1 From e44007e0f97fdae45b73cf61e9962493ddcc6114 Mon Sep 17 00:00:00 2001 From: "Chew, Chiau Ee" Date: Thu, 16 May 2013 15:36:12 +0800 Subject: ALSA: hda - add PCI IDs for Intel BayTrail Add HD Audio Device PCI ID for the Intel BayTrail platform. Signed-off-by: Chew, Chiau Ee Signed-off-by: Artem Bityutskiy Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index de18722c4873..ac75975a4276 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3878,6 +3878,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* BayTrail */ + { PCI_DEVICE(0x8086, 0x0f04), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- cgit v1.2.1 From fd5f940f82cdc0132438a96559b27dd7fd574875 Mon Sep 17 00:00:00 2001 From: Andrew Bresticker Date: Thu, 16 May 2013 12:03:54 -0700 Subject: ASoC: max98090: add digital mic mux to record path The max98090 driver currently treats the digital mic enable as a supply on the record path, causing the digital mic enable to always be turned on when attempting to record. This is incorrect, however, since the digital mic enable is also a mux control where 0 selects the ADC output as input to the record-path DSP and 1 selects the digital mic. This patch adds a virtual DMIC mux to the reocrd path so that we can switch between the ADC and the digital mic for recording. Signed-off-by: Andrew Bresticker Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 22 ++++++++++++++++++---- 1 file changed, 18 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ce0d36412c97..cbb272b1f73d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -857,6 +857,14 @@ static const struct soc_enum mic2_mux_enum = static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); +static const char *dmic_mux_text[] = { "ADC", "DMIC" }; + +static const struct soc_enum dmic_mux_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_mux_text), dmic_mux_text); + +static const struct snd_kcontrol_new max98090_dmic_mux = + SOC_DAPM_ENUM_VIRT("DMIC Mux", dmic_mux_enum); + static const char *max98090_micpre_text[] = { "Off", "On" }; static const struct soc_enum max98090_pa1en_enum = @@ -1144,6 +1152,9 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIC2 Mux", SND_SOC_NOPM, 0, 0, &max98090_mic2_mux), + SND_SOC_DAPM_VIRT_MUX("DMIC Mux", SND_SOC_NOPM, + 0, 0, &max98090_dmic_mux), + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -1336,11 +1347,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"ADCL", NULL, "SHDN"}, {"ADCR", NULL, "SHDN"}, - {"LBENL Mux", "Normal", "ADCL"}, - {"LBENL Mux", "Normal", "DMICL"}, + {"DMIC Mux", "ADC", "ADCL"}, + {"DMIC Mux", "ADC", "ADCR"}, + {"DMIC Mux", "DMIC", "DMICL"}, + {"DMIC Mux", "DMIC", "DMICR"}, + + {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, - {"LBENR Mux", "Normal", "ADCR"}, - {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Normal", "DMIC Mux"}, {"LBENR Mux", "Loopback", "LTENR Mux"}, {"AIFOUTL", NULL, "LBENL Mux"}, -- cgit v1.2.1 From 7d0cd22382f80243e7fce16f6bfc0720d5688370 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 17 May 2013 11:26:15 +0200 Subject: ASoC: simplify registration of snd-soc-dummy device MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4b3be6c3c91e..29b211e9c060 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -159,15 +159,10 @@ int __init snd_soc_util_init(void) { int ret; - soc_dummy_dev = platform_device_alloc("snd-soc-dummy", -1); - if (!soc_dummy_dev) - return -ENOMEM; - - ret = platform_device_add(soc_dummy_dev); - if (ret != 0) { - platform_device_put(soc_dummy_dev); - return ret; - } + soc_dummy_dev = + platform_device_register_simple("snd-soc-dummy", -1, NULL, 0); + if (IS_ERR(soc_dummy_dev)) + return PTR_ERR(soc_dummy_dev); ret = platform_driver_register(&soc_dummy_driver); if (ret != 0) -- cgit v1.2.1 From 571ab6c6f38e44eca40098f58b28e5016c8dbc50 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 15 May 2013 10:09:43 +0300 Subject: ASoC: wm8994: missing break in wm8994_get_fll_config() Smatch complains that: sound/soc/codecs/wm8994.c:2087 wm8994_get_fll_config() warn: missing break? reassigning 'fll->k' Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9f32dd8660d5..303d755b9342 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2080,6 +2080,7 @@ static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, fll->k = K / 10; pr_debug("N=%x K=%x\n", fll->n, fll->k); + break; default: gcd_fll = gcd(freq_out, freq_in); -- cgit v1.2.1 From 435705e89265dec3c641fe75deb748f05e232e59 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 May 2013 11:16:10 -0500 Subject: ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown Acked-by: Vinod Koul Tested-by: Samreen Nilofer --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 303d755b9342..f1c54af45dcf 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2574,17 +2574,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2603,6 +2610,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2641,12 +2649,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2677,6 +2687,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- cgit v1.2.1 From bd1dd8856998408dd72768930958ea2dc84296a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 May 2013 13:29:03 +0100 Subject: ASoC: arizona: Provide simple DAI ops for autoconfiguring interfaces Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 7 +++++++ sound/soc/codecs/arizona.h | 3 ++- 2 files changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 389f23253831..de625813c0e6 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1198,6 +1198,13 @@ const struct snd_soc_dai_ops arizona_dai_ops = { }; EXPORT_SYMBOL_GPL(arizona_dai_ops); +const struct snd_soc_dai_ops arizona_simple_dai_ops = { + .startup = arizona_startup, + .hw_params = arizona_hw_params_rate, + .set_sysclk = arizona_dai_set_sysclk, +}; +EXPORT_SYMBOL_GPL(arizona_simple_dai_ops); + int arizona_init_dai(struct arizona_priv *priv, int id) { struct arizona_dai_priv *dai_priv = &priv->dai[id]; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index af39f1006427..b60b08ccc1d0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -57,7 +57,7 @@ #define ARIZONA_CLK_98MHZ 5 #define ARIZONA_CLK_147MHZ 6 -#define ARIZONA_MAX_DAI 4 +#define ARIZONA_MAX_DAI 6 #define ARIZONA_MAX_ADSP 4 struct arizona; @@ -213,6 +213,7 @@ extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); extern const struct snd_soc_dai_ops arizona_dai_ops; +extern const struct snd_soc_dai_ops arizona_simple_dai_ops; #define ARIZONA_FLL_NAME_LEN 20 -- cgit v1.2.1 From 1804aff60d3bfe34223744ec8c301699bc0b0407 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 May 2013 13:29:04 +0100 Subject: ASoC: wm5102: Stub hookup for Slimbus interface Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 179 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 179 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6e1ee130705a..9a186ff22b55 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -852,6 +852,15 @@ ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); @@ -898,6 +907,15 @@ ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); @@ -1117,6 +1135,56 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, @@ -1188,6 +1256,15 @@ ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"), ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), @@ -1248,6 +1325,14 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "AIF2RX2", "AIF2RX2" }, \ { name, "AIF3RX1", "AIF3RX1" }, \ { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ { name, "EQ1", "EQ1" }, \ { name, "EQ2", "EQ2" }, \ { name, "EQ3", "EQ3" }, \ @@ -1344,13 +1429,41 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF3RX1", NULL, "AIF3 Playback" }, { "AIF3RX2", NULL, "AIF3 Playback" }, + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + { "AIF1 Playback", NULL, "SYSCLK" }, { "AIF2 Playback", NULL, "SYSCLK" }, { "AIF3 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, { "AIF1 Capture", NULL, "SYSCLK" }, { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, { "IN1L PGA", NULL, "IN1L" }, { "IN1R PGA", NULL, "IN1R" }, @@ -1407,6 +1520,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), @@ -1559,6 +1681,63 @@ static struct snd_soc_dai_driver wm5102_dai[] = { .ops = &arizona_dai_ops, .symmetric_rates = 1, }, + { + .name = "wm5102-slim1", + .id = 4, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5102-slim2", + .id = 5, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5102-slim3", + .id = 6, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, }; static int wm5102_codec_probe(struct snd_soc_codec *codec) -- cgit v1.2.1 From d217f9055631fb910f4f2e09ccf6446d93ff6533 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 May 2013 13:29:05 +0100 Subject: ASoC: wm5110: Stub hookup for Slimbus interface Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 179 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 179 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 731884e04776..f53062f8c42e 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -309,6 +309,15 @@ ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); @@ -360,6 +369,15 @@ ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); @@ -549,6 +567,56 @@ SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, @@ -639,6 +707,15 @@ ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"), ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), @@ -689,6 +766,14 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "AIF2RX2", "AIF2RX2" }, \ { name, "AIF3RX1", "AIF3RX1" }, \ { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ { name, "EQ1", "EQ1" }, \ { name, "EQ2", "EQ2" }, \ { name, "EQ3", "EQ3" }, \ @@ -776,13 +861,41 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF3RX1", NULL, "AIF3 Playback" }, { "AIF3RX2", NULL, "AIF3 Playback" }, + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + { "AIF1 Playback", NULL, "SYSCLK" }, { "AIF2 Playback", NULL, "SYSCLK" }, { "AIF3 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, { "AIF1 Capture", NULL, "SYSCLK" }, { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, { "IN1L PGA", NULL, "IN1L" }, { "IN1R PGA", NULL, "IN1R" }, @@ -828,6 +941,15 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), @@ -962,6 +1084,63 @@ static struct snd_soc_dai_driver wm5110_dai[] = { .ops = &arizona_dai_ops, .symmetric_rates = 1, }, + { + .name = "wm5110-slim1", + .id = 4, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5110-slim2", + .id = 5, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm5110-slim3", + .id = 6, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, }; static int wm5110_codec_probe(struct snd_soc_codec *codec) -- cgit v1.2.1 From b296263398f08d21e68d5d7b2afc43228c208b71 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Tue, 21 May 2013 12:04:08 +0200 Subject: ASoC: ab8500-codec: Set tx dai slots from tx_mask Replace hard-coded tx slot numbers from ab8500_codec_set_dai_tdm_slot using the ones requested by the machine driver in tx_mask instead. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 31 +++++++++++++++++++------------ sound/soc/codecs/ab8500-codec.h | 7 +++++++ 2 files changed, 26 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3126cac7b7c8..bace321a83dd 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2236,7 +2236,7 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; - unsigned int val, mask, slots_active; + unsigned int val, mask, slot, slots_active; mask = BIT(AB8500_DIGIFCONF2_IF0WL0) | BIT(AB8500_DIGIFCONF2_IF0WL1); @@ -2292,27 +2292,34 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); /* Setup TDM DA according to active tx slots */ + + if (tx_mask & ~0xff) + return -EINVAL; + mask = AB8500_DASLOTCONFX_SLTODAX_MASK; + tx_mask = tx_mask << AB8500_DA_DATA0_OFFSET; slots_active = hweight32(tx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__, slots_active); + switch (slots_active) { case 0: break; case 1: - /* Slot 9 -> DA_IN1 & DA_IN3 */ - snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + slot = find_first_bit((unsigned long *)&tx_mask, 32); + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; case 2: - /* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */ - snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9); - snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9); - snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); - snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); - + slot = find_first_bit((unsigned long *)&tx_mask, 32); + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); + slot = find_next_bit((unsigned long *)&tx_mask, 32, slot + 1); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; case 8: dev_dbg(dai->codec->dev, diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 114f69a0c629..4224b525cb2f 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -24,6 +24,13 @@ #define AB8500_SUPPORTED_RATE (SNDRV_PCM_RATE_48000) #define AB8500_SUPPORTED_FMT (SNDRV_PCM_FMTBIT_S16_LE) +/* AB8500 interface slot offset definitions */ + +#define AB8500_AD_DATA0_OFFSET 0 +#define AB8500_DA_DATA0_OFFSET 8 +#define AB8500_AD_DATA1_OFFSET 16 +#define AB8500_DA_DATA1_OFFSET 24 + /* AB8500 audio bank (0x0d) register definitions */ #define AB8500_POWERUP 0x00 -- cgit v1.2.1 From da33d723bcb3569400685b4e6e75a9894e2f42a7 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Tue, 21 May 2013 12:04:09 +0200 Subject: ASoC: ab8500-codec: Set rx dai slots from rx_mask Replace hard coded rx slot numbers from ab8500_codec_set_dai_tdm_slot using the ones requested by the machine driver in rx_mask instead. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 29 ++++++++++++++++++++--------- sound/soc/codecs/ab8500-codec.h | 35 +++++++++++++++-------------------- 2 files changed, 35 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index bace321a83dd..4ca45b9d9625 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2334,25 +2334,36 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, } /* Setup TDM AD according to active RX-slots */ + + if (rx_mask & ~0xff) + return -EINVAL; + + rx_mask = rx_mask << AB8500_AD_DATA0_OFFSET; slots_active = hweight32(rx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__, slots_active); + switch (slots_active) { case 0: break; case 1: - /* AD_OUT3 -> slot 0 & 1 */ - snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL, - AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | - AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD); + slot = find_first_bit((unsigned long *)&rx_mask, 32); + snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), + AB8500_MASK_SLOT(slot), + AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); break; case 2: - /* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */ + slot = find_first_bit((unsigned long *)&rx_mask, 32); + snd_soc_update_bits(codec, + AB8500_ADSLOTSEL(slot), + AB8500_MASK_SLOT(slot), + AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); + slot = find_next_bit((unsigned long *)&rx_mask, 32, slot + 1); snd_soc_update_bits(codec, - AB8500_ADSLOTSEL1, - AB8500_MASK_ALL, - AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | - AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD); + AB8500_ADSLOTSEL(slot), + AB8500_MASK_SLOT(slot), + AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT2, slot)); break; case 8: dev_dbg(dai->codec->dev, diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 64c14ce41f69..e2e54425d25e 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -80,6 +80,7 @@ #define AB8500_ADSLOTSEL14 0x2C #define AB8500_ADSLOTSEL15 0x2D #define AB8500_ADSLOTSEL16 0x2E +#define AB8500_ADSLOTSEL(slot) (AB8500_ADSLOTSEL1 + (slot >> 1)) #define AB8500_ADSLOTHIZCTRL1 0x2F #define AB8500_ADSLOTHIZCTRL2 0x30 #define AB8500_ADSLOTHIZCTRL3 0x31 @@ -151,6 +152,7 @@ #define AB8500_CACHEREGNUM (AB8500_LAST_REG + 1) #define AB8500_MASK_ALL 0xFF +#define AB8500_MASK_SLOT(slot) ((slot & 1) ? 0xF0 : 0x0F) #define AB8500_MASK_NONE 0x00 /* AB8500_POWERUP */ @@ -354,28 +356,21 @@ #define AB8500_DIGIFCONF4_IF1WL0 0 /* AB8500_ADSLOTSELX */ -#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x10 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x20 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x30 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x40 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x50 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x60 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x70 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x80 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0xF0 -#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x01 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x02 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x03 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x04 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x05 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x06 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x07 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x08 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0x0F +#define AB8500_AD_OUT1 0x0 +#define AB8500_AD_OUT2 0x1 +#define AB8500_AD_OUT3 0x2 +#define AB8500_AD_OUT4 0x3 +#define AB8500_AD_OUT5 0x4 +#define AB8500_AD_OUT6 0x5 +#define AB8500_AD_OUT7 0x6 +#define AB8500_AD_OUT8 0x7 +#define AB8500_ZEROES 0x8 +#define AB8500_TRISTATE 0xF #define AB8500_ADSLOTSELX_EVEN_SHIFT 0 #define AB8500_ADSLOTSELX_ODD_SHIFT 4 +#define AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(out, slot) \ + ((out) << (((slot) & 1) ? \ + AB8500_ADSLOTSELX_ODD_SHIFT : AB8500_ADSLOTSELX_EVEN_SHIFT)) /* AB8500_ADSLOTHIZCTRL1 */ /* AB8500_ADSLOTHIZCTRL2 */ -- cgit v1.2.1 From 200ceb962f7b00815259bf3cb2df5a0ac15eb99d Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Sat, 18 May 2013 20:25:00 +0800 Subject: ASoC: dfbmcs320: make the driver common for other BT modules DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Signed-off-by: Barry Song Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 +-- sound/soc/codecs/Makefile | 4 +-- sound/soc/codecs/bt-sco.c | 71 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/dfbmcs320.c | 62 --------------------------------- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- 6 files changed, 77 insertions(+), 68 deletions(-) create mode 100644 sound/soc/codecs/bt-sco.c delete mode 100644 sound/soc/codecs/dfbmcs320.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00e31b0..395fda22ade4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9a1f4c..5ba9be87e472 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c new file mode 100644 index 000000000000..a081d9fcb166 --- /dev/null +++ b/sound/soc/codecs/bt-sco.c @@ -0,0 +1,71 @@ +/* + * Driver for generic Bluetooth SCO link + * Copyright 2011 Lars-Peter Clausen + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include + +#include + +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; + +static int bt_sco_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); +} + +static int bt_sco_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { + .driver = { + .name = "bt-sco", + .owner = THIS_MODULE, + }, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, +}; + +module_platform_driver(bt_sco_driver); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c deleted file mode 100644 index 4f4f7f41a7d1..000000000000 --- a/sound/soc/codecs/dfbmcs320.c +++ /dev/null @@ -1,62 +0,0 @@ -/* - * Driver for the DFBM-CS320 bluetooth module - * Copyright 2011 Lars-Peter Clausen - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include -#include -#include - -#include - -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}; - -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; - -static int dfbmcs320_probe(struct platform_device *pdev) -{ - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); -} - -static int dfbmcs320_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); - - return 0; -} - -static struct platform_driver dfmcs320_driver = { - .driver = { - .name = "dfbmcs320", - .owner = THIS_MODULE, - }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, -}; - -module_platform_driver(dfmcs320_driver); - -MODULE_AUTHOR("Lars-Peter Clausen "); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 475fb0d8b3c6..ae0ea87b7d7b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,7 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_DFBMCS320 + select SND_SOC_SCO help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c386917a..807db417d234 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- cgit v1.2.1 From cf1f7c6e8756646db7a0d883013cedd90eb90dd4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 23 May 2013 00:12:53 +0200 Subject: ASoC: Fix early event callback list iteration The power_list field is used when adding a widget to a power sequence list. Use the same field when iterating the list using list_for_each_entry, otherwise we'll see undefined behavior. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 071579be7cb9..35073462d948 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1738,11 +1738,11 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); - list_for_each_entry(w, &down_list, list) { + list_for_each_entry(w, &down_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); } - list_for_each_entry(w, &up_list, list) { + list_for_each_entry(w, &up_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); } -- cgit v1.2.1 From 54d672e922cdb7e015a22a3a2c096dcac5ba284f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:34 +0530 Subject: ALSA: pxa2xx-ac97: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Cc: Eric Miao Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ec54be4efff0..ce431e6e07cf 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -230,7 +230,6 @@ static int pxa2xx_ac97_remove(struct platform_device *dev) if (card) { snd_card_free(card); - platform_set_drvdata(dev, NULL); pxa2xx_ac97_hw_remove(dev); } -- cgit v1.2.1 From 5ed5824bf43da9e2931fb94043a2942d73bec283 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:35 +0530 Subject: ALSA: aloop: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Cc: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 6f78de9c6fb6..f7589923effa 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1183,7 +1183,6 @@ static int loopback_probe(struct platform_device *devptr) static int loopback_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From 186c1821f94719a02e570cf45e2047ffb6ceaddf Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:36 +0530 Subject: ALSA: ml403-ac97cr: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/drivers/ml403-ac97cr.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 8125a7e95ee4..95ea4a153ea4 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1325,7 +1325,6 @@ static int snd_ml403_ac97cr_probe(struct platform_device *pfdev) static int snd_ml403_ac97cr_remove(struct platform_device *pfdev) { snd_card_free(platform_get_drvdata(pfdev)); - platform_set_drvdata(pfdev, NULL); return 0; } -- cgit v1.2.1 From 984ef60cb148c468925f5b1a15a3f425c9d6bf6c Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:37 +0530 Subject: ALSA: mpu401: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/drivers/mpu401/mpu401.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index da1a29bfc85d..90a3a7b38a2a 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -129,7 +129,6 @@ static int snd_mpu401_probe(struct platform_device *devptr) static int snd_mpu401_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From a204341dae68bee125f4e7f22c5640cb4e3aae16 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:38 +0530 Subject: ALSA: mtpav: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/drivers/mtpav.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 9f1815b99a15..e5ec7eb27dec 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -749,7 +749,6 @@ static int snd_mtpav_probe(struct platform_device *dev) static int snd_mtpav_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From a4302ede92d143dd6c0f3dfbe25bcce852249e61 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:39 +0530 Subject: ALSA: pcsp: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Cc: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 7a5fdb9b0afc..1c19cd7ad26e 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -189,7 +189,6 @@ static int pcsp_remove(struct platform_device *dev) struct snd_pcsp *chip = platform_get_drvdata(dev); alsa_card_pcsp_exit(chip); pcspkr_input_remove(chip->input_dev); - platform_set_drvdata(dev, NULL); return 0; } -- cgit v1.2.1 From 45837cb214a7fd076ae66a87a9425588156d3a78 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:40 +0530 Subject: ALSA: serial-u16550: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/drivers/serial-u16550.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 7425dd8c1f09..e0bf5e77b43a 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -985,7 +985,6 @@ static int snd_serial_probe(struct platform_device *devptr) static int snd_serial_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From 50c7d0da64e031c072fdc51b30d05c98761e3ebc Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:41 +0530 Subject: ALSA: virmidi: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/drivers/virmidi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index cc4be88d7318..ace3879e8d96 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -132,7 +132,6 @@ static int snd_virmidi_probe(struct platform_device *devptr) static int snd_virmidi_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From 464ede3ce59ba6144ff117ffa6427cac77fc6807 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:42 +0530 Subject: ALSA: powermac: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/ppc/powermac.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 09fc848d32ec..8abb521b4814 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -139,7 +139,6 @@ __error: static int snd_pmac_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From 2bc594a2764983532887c2c606172bd262c60644 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:43 +0530 Subject: ALSA: sh: aica: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Cc: Adrian McMenamin Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/sh/aica.c b/sound/sh/aica.c index e59a73a9bc42..78a369785a9e 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -598,7 +598,6 @@ static int snd_aica_remove(struct platform_device *devptr) return -ENODEV; snd_card_free(dreamcastcard->card); kfree(dreamcastcard); - platform_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From 3cf981484a002fd02c410741d30b234227f89568 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 20 May 2013 14:08:44 +0530 Subject: ALSA: sh_dac_audio: Remove redundant platform_set_drvdata() Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat Cc: Rafael Ignacio Zurita Signed-off-by: Takashi Iwai --- sound/sh/sh_dac_audio.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index e68c4fc91a03..7c9422c4fc0f 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -290,8 +290,6 @@ static int snd_sh_dac_pcm(struct snd_sh_dac *chip, int device) static int snd_sh_dac_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; } -- cgit v1.2.1 From 47f5b692e04a8d7989ee14591a61be26e340a17b Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 23 May 2013 13:58:00 +0200 Subject: ASoC: adau1701: refactor firmware loading function Pass a struct i2c_client * to adau1701_load_firmware directly to make the code more readable. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index dafdbe87edeb..95e1677665e9 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -180,9 +180,9 @@ static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) return value; } -static int adau1701_load_firmware(struct snd_soc_codec *codec) +static int adau1701_load_firmware(struct i2c_client *client) { - return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); + return process_sigma_firmware(client, ADAU1701_FIRMWARE); } static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, @@ -455,10 +455,11 @@ static struct snd_soc_dai_driver adau1701_dai = { static int adau1701_probe(struct snd_soc_codec *codec) { int ret; + struct i2c_client *client = to_i2c_client(codec->dev); - codec->control_data = to_i2c_client(codec->dev); + codec->control_data = client; - ret = adau1701_load_firmware(codec); + ret = adau1701_load_firmware(client); if (ret) dev_warn(codec->dev, "Failed to load firmware\n"); -- cgit v1.2.1 From e6c2e7eb27fc512af6875d7f2cf313e29c61be0b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 24 May 2013 15:18:10 +0200 Subject: ALSA: Constify the snd_pcm_substream struct ops field The ops field of the snd_pcm_substream struct is never modified inside the ALSA core. Making it const allows drivers to declare their snd_pcm_ops struct as const. Signed-off-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 41b3dfe68698..82bb029d4414 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -568,7 +568,8 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) * * Sets the given PCM operators to the pcm instance. */ -void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, struct snd_pcm_ops *ops) +void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, + const struct snd_pcm_ops *ops) { struct snd_pcm_str *stream = &pcm->streams[direction]; struct snd_pcm_substream *substream; -- cgit v1.2.1 From 8edbb198a62e2c3d0bea06ce50a4d45a009849b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 May 2013 16:30:39 +0200 Subject: ALSA: Fix the default suffix string with high card number ALSA core tries to add a suffix as "_1" automatically when the given id string conflicts. The current code assumes implicitly that the max card number is 16 so that the single hex "_X" suffix can be put. However, with the dynamic device management, the card can be at most 32, so it can put even a non-hex character there. Also, when the max card number is increased in future, this would result in worse. This patch rewrites the code to add the suffix string in a simpler (thus cleaner) way. It can support up to three digits, so it should suffice for most requirements. Signed-off-by: Takashi Iwai --- sound/core/init.c | 30 +++++++++++++----------------- 1 file changed, 13 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 6ef06400dfc8..ed4a4811b6a1 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -549,7 +549,6 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, const char *nid) { int len, loops; - bool with_suffix; bool is_default = false; char *id; @@ -565,26 +564,23 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, is_default = true; } - with_suffix = false; + len = strlen(id); for (loops = 0; loops < SNDRV_CARDS; loops++) { + char *spos; + char sfxstr[5]; /* "_012" */ + int sfxlen; + if (card_id_ok(card, id)) return; /* OK */ - len = strlen(id); - if (!with_suffix) { - /* add the "_X" suffix */ - char *spos = id + len; - if (len > sizeof(card->id) - 3) - spos = id + sizeof(card->id) - 3; - strcpy(spos, "_1"); - with_suffix = true; - } else { - /* modify the existing suffix */ - if (id[len - 1] != '9') - id[len - 1]++; - else - id[len - 1] = 'A'; - } + /* Add _XYZ suffix */ + sprintf(sfxstr, "_%X", loops + 1); + sfxlen = strlen(sfxstr); + if (len + sfxlen >= sizeof(card->id)) + spos = id + sizeof(card->id) - sfxlen - 1; + else + spos = id + len; + strcpy(spos, sfxstr); } /* fallback to the default id */ if (!is_default) { -- cgit v1.2.1 From 7bb2491b35a254fe6fd592c32a142a2f2f31fe6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 May 2013 08:46:39 +0200 Subject: ALSA: Add kconfig to specify the max card numbers Currently ALSA supports up to 32 card instances when the dynamic minor is used. While 32 cards are usually big enough for normal use cases, there are sometimes weird requirements with more card support. Actually, this limitation, 32, comes from the index option, where you can pass the bit mask to assign the card. Other than that, we can actually give more cards up to the minor number limits (currently 256, which can be extended more, too). This patch adds a new Kconfig to specify the max card numbers, and changes a few places to accept more than 32 cards. The only incompatibility with high card numbers would be the handling of index option. The index option can be still used to pass the bitmask for card assignments, but this works only up to 32 slots. More than 32, no bitmask style option is available but only a single slot can be specified via index option. Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 9 +++++++++ sound/core/init.c | 25 ++++++++++++++++--------- 2 files changed, 25 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index b413ed05e74d..c0c2f57a0d6f 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -157,6 +157,15 @@ config SND_DYNAMIC_MINORS If you are unsure about this, say N here. +config SND_MAX_CARDS + int "Max number of sound cards" + range 4 256 + default 32 + depends on SND_DYNAMIC_MINORS + help + Specify the max number of sound cards that can be assigned + on a single machine. + config SND_SUPPORT_OLD_API bool "Support old ALSA API" default y diff --git a/sound/core/init.c b/sound/core/init.c index ed4a4811b6a1..6b9087115da2 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -46,7 +46,8 @@ static LIST_HEAD(shutdown_files); static const struct file_operations snd_shutdown_f_ops; -static unsigned int snd_cards_lock; /* locked for registering/using */ +/* locked for registering/using */ +static DECLARE_BITMAP(snd_cards_lock, SNDRV_CARDS); struct snd_card *snd_cards[SNDRV_CARDS]; EXPORT_SYMBOL(snd_cards); @@ -167,29 +168,35 @@ int snd_card_create(int idx, const char *xid, err = 0; mutex_lock(&snd_card_mutex); if (idx < 0) { - for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) + for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { /* idx == -1 == 0xffff means: take any free slot */ - if (~snd_cards_lock & idx & 1<= SNDRV_CARDS) err = -ENODEV; @@ -199,7 +206,7 @@ int snd_card_create(int idx, const char *xid, idx, snd_ecards_limit - 1, err); goto __error; } - snd_cards_lock |= 1 << idx; /* lock it */ + set_bit(idx, snd_cards_lock); /* lock it */ if (idx >= snd_ecards_limit) snd_ecards_limit = idx + 1; /* increase the limit */ mutex_unlock(&snd_card_mutex); @@ -249,7 +256,7 @@ int snd_card_locked(int card) int locked; mutex_lock(&snd_card_mutex); - locked = snd_cards_lock & (1 << card); + locked = test_bit(card, snd_cards_lock); mutex_unlock(&snd_card_mutex); return locked; } @@ -361,7 +368,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 1: disable fops (user space) operations for ALSA API */ mutex_lock(&snd_card_mutex); snd_cards[card->number] = NULL; - snd_cards_lock &= ~(1 << card->number); + clear_bit(card->number, snd_cards_lock); mutex_unlock(&snd_card_mutex); /* phase 2: replace file->f_op with special dummy operations */ -- cgit v1.2.1 From b6b5e76bb8bb22ecff90a7840dc4845d63328289 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 23 May 2013 20:14:50 +0200 Subject: ASoC: Add ssm2518 support This patch adds a ASoC CODEC driver for the SSM2516. The SSM2516 is a stereo Class-D audio amplifier with an I2S interface for audio in and a built-in dynamic range control processor. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2518.c | 856 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2518.h | 20 ++ 4 files changed, 882 insertions(+) create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00e31b0..d76609adb85b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF + select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C @@ -313,6 +314,9 @@ config SND_SOC_SN95031 config SND_SOC_SPDIF tristate +config SND_SOC_SSM2518 + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9a1f4c..d85be48a6c07 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transciever.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o @@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c new file mode 100644 index 000000000000..3139a1bde295 --- /dev/null +++ b/sound/soc/codecs/ssm2518.c @@ -0,0 +1,856 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ssm2518.h" + +#define SSM2518_REG_POWER1 0x00 +#define SSM2518_REG_CLOCK 0x01 +#define SSM2518_REG_SAI_CTRL1 0x02 +#define SSM2518_REG_SAI_CTRL2 0x03 +#define SSM2518_REG_CHAN_MAP 0x04 +#define SSM2518_REG_LEFT_VOL 0x05 +#define SSM2518_REG_RIGHT_VOL 0x06 +#define SSM2518_REG_MUTE_CTRL 0x07 +#define SSM2518_REG_FAULT_CTRL 0x08 +#define SSM2518_REG_POWER2 0x09 +#define SSM2518_REG_DRC_1 0x0a +#define SSM2518_REG_DRC_2 0x0b +#define SSM2518_REG_DRC_3 0x0c +#define SSM2518_REG_DRC_4 0x0d +#define SSM2518_REG_DRC_5 0x0e +#define SSM2518_REG_DRC_6 0x0f +#define SSM2518_REG_DRC_7 0x10 +#define SSM2518_REG_DRC_8 0x11 +#define SSM2518_REG_DRC_9 0x12 + +#define SSM2518_POWER1_RESET BIT(7) +#define SSM2518_POWER1_NO_BCLK BIT(5) +#define SSM2518_POWER1_MCS_MASK (0xf << 1) +#define SSM2518_POWER1_MCS_64FS (0x0 << 1) +#define SSM2518_POWER1_MCS_128FS (0x1 << 1) +#define SSM2518_POWER1_MCS_256FS (0x2 << 1) +#define SSM2518_POWER1_MCS_384FS (0x3 << 1) +#define SSM2518_POWER1_MCS_512FS (0x4 << 1) +#define SSM2518_POWER1_MCS_768FS (0x5 << 1) +#define SSM2518_POWER1_MCS_100FS (0x6 << 1) +#define SSM2518_POWER1_MCS_200FS (0x7 << 1) +#define SSM2518_POWER1_MCS_400FS (0x8 << 1) +#define SSM2518_POWER1_SPWDN BIT(0) + +#define SSM2518_CLOCK_ASR BIT(0) + +#define SSM2518_SAI_CTRL1_FMT_MASK (0x3 << 5) +#define SSM2518_SAI_CTRL1_FMT_I2S (0x0 << 5) +#define SSM2518_SAI_CTRL1_FMT_LJ (0x1 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_24BIT (0x2 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_16BIT (0x3 << 5) + +#define SSM2518_SAI_CTRL1_SAI_MASK (0x7 << 2) +#define SSM2518_SAI_CTRL1_SAI_I2S (0x0 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_2 (0x1 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_4 (0x2 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_8 (0x3 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_16 (0x4 << 2) +#define SSM2518_SAI_CTRL1_SAI_MONO (0x5 << 2) + +#define SSM2518_SAI_CTRL1_FS_MASK (0x3) +#define SSM2518_SAI_CTRL1_FS_8000_12000 (0x0) +#define SSM2518_SAI_CTRL1_FS_16000_24000 (0x1) +#define SSM2518_SAI_CTRL1_FS_32000_48000 (0x2) +#define SSM2518_SAI_CTRL1_FS_64000_96000 (0x3) + +#define SSM2518_SAI_CTRL2_BCLK_INTERAL BIT(7) +#define SSM2518_SAI_CTRL2_LRCLK_PULSE BIT(6) +#define SSM2518_SAI_CTRL2_LRCLK_INVERT BIT(5) +#define SSM2518_SAI_CTRL2_MSB BIT(4) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK (0x3 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_32 (0x0 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_24 (0x1 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_16 (0x2 << 2) +#define SSM2518_SAI_CTRL2_BCLK_INVERT BIT(1) + +#define SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET 4 +#define SSM2518_CHAN_MAP_RIGHT_SLOT_MASK 0xf0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET 0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_MASK 0x0f + +#define SSM2518_MUTE_CTRL_ANA_GAIN BIT(5) +#define SSM2518_MUTE_CTRL_MUTE_MASTER BIT(0) + +#define SSM2518_POWER2_APWDN BIT(0) + +#define SSM2518_DAC_MUTE BIT(6) +#define SSM2518_DAC_FS_MASK 0x07 +#define SSM2518_DAC_FS_8000 0x00 +#define SSM2518_DAC_FS_16000 0x01 +#define SSM2518_DAC_FS_32000 0x02 +#define SSM2518_DAC_FS_64000 0x03 +#define SSM2518_DAC_FS_128000 0x04 + +struct ssm2518 { + struct regmap *regmap; + bool right_j; + + unsigned int sysclk; + const struct snd_pcm_hw_constraint_list *constraints; + + int enable_gpio; +}; + +static const struct reg_default ssm2518_reg_defaults[] = { + { 0x00, 0x05 }, + { 0x01, 0x00 }, + { 0x02, 0x02 }, + { 0x03, 0x00 }, + { 0x04, 0x10 }, + { 0x05, 0x40 }, + { 0x06, 0x40 }, + { 0x07, 0x81 }, + { 0x08, 0x0c }, + { 0x09, 0x99 }, + { 0x0a, 0x7c }, + { 0x0b, 0x5b }, + { 0x0c, 0x57 }, + { 0x0d, 0x89 }, + { 0x0e, 0x8c }, + { 0x0f, 0x77 }, + { 0x10, 0x26 }, + { 0x11, 0x1c }, + { 0x12, 0x97 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm2518_vol_tlv, -7125, 2400); +static const DECLARE_TLV_DB_SCALE(ssm2518_compressor_tlv, -3400, 200, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_expander_tlv, -8100, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_noise_gate_tlv, -9600, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_post_drc_tlv, -2400, 300, 0); + +static const DECLARE_TLV_DB_RANGE(ssm2518_limiter_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 7, 15, TLV_DB_SCALE_ITEM(-800, 100, 0), +); + +static const char * const ssm2518_drc_peak_detector_attack_time_text[] = { + "0 ms", "0.1 ms", "0.19 ms", "0.37 ms", "0.75 ms", "1.5 ms", "3 ms", + "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", "192 ms", "384 ms", + "768 ms", "1536 ms", +}; + +static const char * const ssm2518_drc_peak_detector_release_time_text[] = { + "0 ms", "1.5 ms", "3 ms", "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", + "192 ms", "384 ms", "768 ms", "1536 ms", "3072 ms", "6144 ms", + "12288 ms", "24576 ms" +}; + +static const char * const ssm2518_drc_hold_time_text[] = { + "0 ms", "0.67 ms", "1.33 ms", "2.67 ms", "5.33 ms", "10.66 ms", + "21.32 ms", "42.64 ms", "85.28 ms", "170.56 ms", "341.12 ms", + "682.24 ms", "1364 ms", +}; + +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, + SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, + SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, + SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, + SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, + SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, + SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, + SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); + +static const struct snd_kcontrol_new ssm2518_snd_controls[] = { + SOC_SINGLE("Playback De-emphasis Switch", SSM2518_REG_MUTE_CTRL, + 4, 1, 0), + SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2518_REG_LEFT_VOL, + SSM2518_REG_RIGHT_VOL, 0, 0xff, 1, ssm2518_vol_tlv), + SOC_DOUBLE("Master Playback Switch", SSM2518_REG_MUTE_CTRL, 2, 1, 1, 1), + + SOC_SINGLE("Amp Low Power Mode Switch", SSM2518_REG_POWER2, 4, 1, 0), + SOC_SINGLE("DAC Low Power Mode Switch", SSM2518_REG_POWER2, 3, 1, 0), + + SOC_SINGLE("DRC Limiter Switch", SSM2518_REG_DRC_1, 5, 1, 0), + SOC_SINGLE("DRC Compressor Switch", SSM2518_REG_DRC_1, 4, 1, 0), + SOC_SINGLE("DRC Expander Switch", SSM2518_REG_DRC_1, 3, 1, 0), + SOC_SINGLE("DRC Noise Gate Switch", SSM2518_REG_DRC_1, 2, 1, 0), + SOC_DOUBLE("DRC Switch", SSM2518_REG_DRC_1, 0, 1, 1, 0), + + SOC_SINGLE_TLV("DRC Limiter Threshold Volume", + SSM2518_REG_DRC_3, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Compressor Lower Threshold Volume", + SSM2518_REG_DRC_3, 0, 15, 1, ssm2518_compressor_tlv), + SOC_SINGLE_TLV("DRC Expander Upper Threshold Volume", SSM2518_REG_DRC_4, + 4, 15, 1, ssm2518_expander_tlv), + SOC_SINGLE_TLV("DRC Noise Gate Threshold Volume", + SSM2518_REG_DRC_4, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Upper Output Threshold Volume", + SSM2518_REG_DRC_5, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Lower Output Threshold Volume", + SSM2518_REG_DRC_5, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Post Volume", SSM2518_REG_DRC_8, + 2, 15, 1, ssm2518_post_drc_tlv), + + SOC_ENUM("DRC Peak Detector Attack Time", + ssm2518_drc_peak_detector_attack_time_enum), + SOC_ENUM("DRC Peak Detector Release Time", + ssm2518_drc_peak_detector_release_time_enum), + SOC_ENUM("DRC Attack Time", ssm2518_drc_attack_time_enum), + SOC_ENUM("DRC Decay Time", ssm2518_drc_decay_time_enum), + SOC_ENUM("DRC Hold Time", ssm2518_drc_hold_time_enum), + SOC_ENUM("DRC Noise Gate Hold Time", + ssm2518_drc_noise_gate_hold_time_enum), + SOC_ENUM("DRC RMS Averaging Time", ssm2518_drc_rms_averaging_time_enum), +}; + +static const struct snd_soc_dapm_widget ssm2518_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SSM2518_REG_POWER2, 1, 1), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SSM2518_REG_POWER2, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route ssm2518_routes[] = { + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +struct ssm2518_mcs_lut { + unsigned int rate; + const unsigned int *sysclks; +}; + +static const unsigned int ssm2518_sysclks_2048000[] = { + 2048000, 4096000, 8192000, 12288000, 16384000, 24576000, + 3200000, 6400000, 12800000, 0 +}; + +static const unsigned int ssm2518_sysclks_2822000[] = { + 2822000, 5644800, 11289600, 16934400, 22579200, 33868800, + 4410000, 8820000, 17640000, 0 +}; + +static const unsigned int ssm2518_sysclks_3072000[] = { + 3072000, 6144000, 12288000, 16384000, 24576000, 38864000, + 4800000, 9600000, 19200000, 0 +}; + +static const struct ssm2518_mcs_lut ssm2518_mcs_lut[] = { + { 8000, ssm2518_sysclks_2048000, }, + { 11025, ssm2518_sysclks_2822000, }, + { 12000, ssm2518_sysclks_3072000, }, + { 16000, ssm2518_sysclks_2048000, }, + { 24000, ssm2518_sysclks_3072000, }, + { 22050, ssm2518_sysclks_2822000, }, + { 32000, ssm2518_sysclks_2048000, }, + { 44100, ssm2518_sysclks_2822000, }, + { 48000, ssm2518_sysclks_3072000, }, + { 96000, ssm2518_sysclks_3072000, }, +}; + +static const unsigned int ssm2518_rates_2048000[] = { + 8000, 16000, 32000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2048000 = { + .list = ssm2518_rates_2048000, + .count = ARRAY_SIZE(ssm2518_rates_2048000), +}; + +static const unsigned int ssm2518_rates_2822000[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2822000 = { + .list = ssm2518_rates_2822000, + .count = ARRAY_SIZE(ssm2518_rates_2822000), +}; + +static const unsigned int ssm2518_rates_3072000[] = { + 12000, 24000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_3072000 = { + .list = ssm2518_rates_3072000, + .count = ARRAY_SIZE(ssm2518_rates_3072000), +}; + +static const unsigned int ssm2518_rates_12288000[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { + .list = ssm2518_rates_12288000, + .count = ARRAY_SIZE(ssm2518_rates_12288000), +}; + +static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, + unsigned int rate) +{ + const unsigned int *sysclks = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(ssm2518_mcs_lut); i++) { + if (ssm2518_mcs_lut[i].rate == rate) { + sysclks = ssm2518_mcs_lut[i].sysclks; + break; + } + } + + if (!sysclks) + return -EINVAL; + + for (i = 0; sysclks[i]; i++) { + if (sysclks[i] == ssm2518->sysclk) + return i; + } + + return -EINVAL; +} + +static int ssm2518_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int ctrl1, ctrl1_mask; + int mcs; + int ret; + + mcs = ssm2518_lookup_mcs(ssm2518, rate); + if (mcs < 0) + return mcs; + + ctrl1_mask = SSM2518_SAI_CTRL1_FS_MASK; + + if (rate >= 8000 && rate <= 12000) + ctrl1 = SSM2518_SAI_CTRL1_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + ctrl1 = SSM2518_SAI_CTRL1_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + ctrl1 = SSM2518_SAI_CTRL1_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + ctrl1 = SSM2518_SAI_CTRL1_FS_64000_96000; + else + return -EINVAL; + + if (ssm2518->right_j) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl1_mask |= SSM2518_SAI_CTRL1_FMT_MASK; + } + + /* Disable auto samplerate detection */ + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_CLOCK, + SSM2518_CLOCK_ASR, SSM2518_CLOCK_ASR); + if (ret < 0) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + ctrl1_mask, ctrl1); + if (ret < 0) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_MCS_MASK, mcs << 1); +} + +static int ssm2518_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = SSM2518_MUTE_CTRL_MUTE_MASTER; + else + val = 0; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_MUTE_CTRL, + SSM2518_MUTE_CTRL_MUTE_MASTER, val); +} + +static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1 = 0, ctrl2 = 0; + bool invert_fclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + ssm2518->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + ssm2518->right_j = true; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = false; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_INVERT; + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, ctrl1); + if (ret) + return ret; + + return regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, ctrl2); +} + +static int ssm2518_set_power(struct ssm2518 *ssm2518, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN, SSM2518_POWER1_SPWDN); + regcache_mark_dirty(ssm2518->regmap); + } + + if (gpio_is_valid(ssm2518->enable_gpio)) + gpio_set_value(ssm2518->enable_gpio, enable); + + regcache_cache_only(ssm2518->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN | SSM2518_POWER1_RESET, 0x00); + regcache_sync(ssm2518->regmap); + } + + return ret; +} + +static int ssm2518_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm2518_set_power(ssm2518, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm2518_set_power(ssm2518, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1, ctrl2; + int left_slot, right_slot; + int ret; + + if (slots == 0) + return regmap_update_bits(ssm2518->regmap, + SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, + SSM2518_SAI_CTRL1_SAI_I2S); + + if (tx_mask == 0 || tx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + /* We assume the left channel < right channel */ + left_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (width) { + case 16: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_16; + break; + case 24: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_24; + break; + case 32: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 1: + ctrl1 = SSM2518_SAI_CTRL1_SAI_MONO; + break; + case 2: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_2; + break; + case 4: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_4; + break; + case 8: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_8; + break; + case 16: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_CHAN_MAP, + (left_slot << SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET) | + (right_slot << SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + SSM2518_SAI_CTRL1_SAI_MASK, ctrl1); + if (ret) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, + SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK, ctrl2); +} + +static int ssm2518_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + + if (ssm2518->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, ssm2518->constraints); + + return 0; +} + +#define SSM2518_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm2518_dai_ops = { + .startup = ssm2518_startup, + .hw_params = ssm2518_hw_params, + .digital_mute = ssm2518_mute, + .set_fmt = ssm2518_set_dai_fmt, + .set_tdm_slot = ssm2518_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ssm2518_dai = { + .name = "ssm2518-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SSM2518_FORMATS, + }, + .ops = &ssm2518_dai_ops, +}; + +static int ssm2518_probe(struct snd_soc_codec *codec) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = ssm2518->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int ssm2518_remove(struct snd_soc_codec *codec) +{ + ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (clk_id != SSM2518_SYSCLK) + return -EINVAL; + + switch (source) { + case SSM2518_SYSCLK_SRC_MCLK: + val = 0; + break; + case SSM2518_SYSCLK_SRC_BCLK: + /* In this case the bitclock is used as the system clock, and + * the bitclock signal needs to be connected to the MCLK pin and + * the BCLK pin is left unconnected */ + val = SSM2518_POWER1_NO_BCLK; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 0: + ssm2518->constraints = NULL; + break; + case 2048000: + case 4096000: + case 8192000: + case 3200000: + case 6400000: + case 12800000: + ssm2518->constraints = &ssm2518_constraints_2048000; + break; + case 2822000: + case 5644800: + case 11289600: + case 16934400: + case 22579200: + case 33868800: + case 4410000: + case 8820000: + case 17640000: + ssm2518->constraints = &ssm2518_constraints_2822000; + break; + case 3072000: + case 6144000: + case 38864000: + case 4800000: + case 9600000: + case 19200000: + ssm2518->constraints = &ssm2518_constraints_3072000; + break; + case 12288000: + case 16384000: + case 24576000: + ssm2518->constraints = &ssm2518_constraints_12288000; + break; + default: + return -EINVAL; + } + + ssm2518->sysclk = freq; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_NO_BCLK, val); +} + +static struct snd_soc_codec_driver ssm2518_codec_driver = { + .probe = ssm2518_probe, + .remove = ssm2518_remove, + .set_bias_level = ssm2518_set_bias_level, + .set_sysclk = ssm2518_set_sysclk, + .idle_bias_off = true, + + .controls = ssm2518_snd_controls, + .num_controls = ARRAY_SIZE(ssm2518_snd_controls), + .dapm_widgets = ssm2518_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2518_dapm_widgets), + .dapm_routes = ssm2518_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), +}; + +static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config ssm2518_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM2518_REG_DRC_9, + .volatile_reg = ssm2518_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm2518_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm2518_reg_defaults), +}; + +static int ssm2518_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2518_platform_data *pdata = i2c->dev.platform_data; + struct ssm2518 *ssm2518; + int ret; + + ssm2518 = devm_kzalloc(&i2c->dev, sizeof(*ssm2518), GFP_KERNEL); + if (ssm2518 == NULL) + return -ENOMEM; + + if (pdata) { + ssm2518->enable_gpio = pdata->enable_gpio; + } else if (i2c->dev.of_node) { + ssm2518->enable_gpio = of_get_gpio(i2c->dev.of_node, 0); + if (ssm2518->enable_gpio < 0 && ssm2518->enable_gpio != -ENOENT) + return ssm2518->enable_gpio; + } else { + ssm2518->enable_gpio = -1; + } + + if (gpio_is_valid(ssm2518->enable_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ssm2518->enable_gpio, + GPIOF_OUT_INIT_HIGH, "SSM2518 nSD"); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, ssm2518); + + ssm2518->regmap = devm_regmap_init_i2c(i2c, &ssm2518_regmap_config); + if (IS_ERR(ssm2518->regmap)) + return PTR_ERR(ssm2518->regmap); + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(ssm2518->regmap, true); + ret = regmap_write(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_RESET); + regcache_cache_bypass(ssm2518->regmap, false); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER2, + SSM2518_POWER2_APWDN, 0x00); + if (ret) + return ret; + + ret = ssm2518_set_power(ssm2518, false); + if (ret) + return ret; + + return snd_soc_register_codec(&i2c->dev, &ssm2518_codec_driver, + &ssm2518_dai, 1); +} + +static int ssm2518_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2518_i2c_ids[] = { + { "ssm2518", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); + +static struct i2c_driver ssm2518_driver = { + .driver = { + .name = "ssm2518", + .owner = THIS_MODULE, + }, + .probe = ssm2518_i2c_probe, + .remove = ssm2518_i2c_remove, + .id_table = ssm2518_i2c_ids, +}; +module_i2c_driver(ssm2518_driver); + +MODULE_DESCRIPTION("ASoC SSM2518 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2518.h b/sound/soc/codecs/ssm2518.h new file mode 100644 index 000000000000..62511d80518e --- /dev/null +++ b/sound/soc/codecs/ssm2518.h @@ -0,0 +1,20 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#ifndef __SND_SOC_CODECS_SSM2518_H__ +#define __SND_SOC_CODECS_SSM2518_H__ + +#define SSM2518_SYSCLK 0 + +enum ssm2518_sysclk_src { + SSM2518_SYSCLK_SRC_MCLK = 0, + SSM2518_SYSCLK_SRC_BCLK = 1, +}; + +#endif -- cgit v1.2.1 From 04561eacaa6ccd1988e468cdcbf4acc475ae2221 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 23 May 2013 15:46:05 +0200 Subject: ASoC: codecs: adau1701: add DT bindings Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 95e1677665e9..3fc176387351 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -13,6 +13,9 @@ #include #include #include +#include +#include +#include #include #include #include @@ -452,6 +455,14 @@ static struct snd_soc_dai_driver adau1701_dai = { .symmetric_rates = 1, }; +#ifdef CONFIG_OF +static const struct of_device_id adau1701_dt_ids[] = { + { .compatible = "adi,adau1701", }, + { } +}; +MODULE_DEVICE_TABLE(of, adau1701_dt_ids); +#endif + static int adau1701_probe(struct snd_soc_codec *codec) { int ret; @@ -494,12 +505,33 @@ static int adau1701_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct adau1701 *adau1701; + struct device *dev = &client->dev; + int gpio_nreset = -EINVAL; int ret; - adau1701 = devm_kzalloc(&client->dev, sizeof(*adau1701), GFP_KERNEL); + adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); if (!adau1701) return -ENOMEM; + if (dev->of_node) { + gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); + if (gpio_nreset < 0 && gpio_nreset != -ENOENT) + return gpio_nreset; + } + + if (gpio_is_valid(gpio_nreset)) { + ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW, + "ADAU1701 Reset"); + if (ret < 0) + return ret; + + /* minimum reset time is 20ns */ + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* power-up time may be as long as 85ms */ + mdelay(85); + } + i2c_set_clientdata(client, adau1701); ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); @@ -522,6 +554,7 @@ static struct i2c_driver adau1701_i2c_driver = { .driver = { .name = "adau1701", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(adau1701_dt_ids), }, .probe = adau1701_i2c_probe, .remove = adau1701_i2c_remove, -- cgit v1.2.1 From 6ee0b4b0ef871632b067f216b3032bf8db93c510 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Fri, 24 May 2013 12:39:15 +0200 Subject: ASoC: ux500: Drop pinctrl sleep support Drop pinctrl default/sleep state switching code, as it was breaking the capture interface by putting the I2S pins in hi-z mode regardless of its usage status, and not giving any real benefit. Pinctrl default mode configuration is already managed automatically by a specific pinctrl hog. Signed-off-by: Fabio Baltieri Acked-by: Linus Walleij Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.c | 56 ++--------------------------------------- sound/soc/ux500/ux500_msp_i2s.h | 6 ----- 2 files changed, 2 insertions(+), 60 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index f2db6c90a9e2..b029b2d673d7 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -15,7 +15,6 @@ #include #include -#include #include #include #include @@ -26,9 +25,6 @@ #include "ux500_msp_i2s.h" -/* MSP1/3 Tx/Rx usage protection */ -static DEFINE_SPINLOCK(msp_rxtx_lock); - /* Protocol desciptors */ static const struct msp_protdesc prot_descs[] = { { /* I2S */ @@ -356,24 +352,8 @@ static int configure_multichannel(struct ux500_msp *msp, static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) { - int status = 0, retval = 0; + int status = 0; u32 reg_val_DMACR, reg_val_GCR; - unsigned long flags; - - /* Check msp state whether in RUN or CONFIGURED Mode */ - if (msp->msp_state == MSP_STATE_IDLE) { - spin_lock_irqsave(&msp_rxtx_lock, flags); - if (msp->pinctrl_rxtx_ref == 0 && - !(IS_ERR(msp->pinctrl_p) || IS_ERR(msp->pinctrl_def))) { - retval = pinctrl_select_state(msp->pinctrl_p, - msp->pinctrl_def); - if (retval) - pr_err("could not set MSP defstate\n"); - } - if (!retval) - msp->pinctrl_rxtx_ref++; - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - } /* Configure msp with protocol dependent settings */ configure_protocol(msp, config); @@ -630,8 +610,7 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) { - int status = 0, retval = 0; - unsigned long flags; + int status = 0; dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir); @@ -643,18 +622,6 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) (~(FRAME_GEN_ENABLE | SRG_ENABLE))), msp->registers + MSP_GCR); - spin_lock_irqsave(&msp_rxtx_lock, flags); - WARN_ON(!msp->pinctrl_rxtx_ref); - msp->pinctrl_rxtx_ref--; - if (msp->pinctrl_rxtx_ref == 0 && - !(IS_ERR(msp->pinctrl_p) || IS_ERR(msp->pinctrl_sleep))) { - retval = pinctrl_select_state(msp->pinctrl_p, - msp->pinctrl_sleep); - if (retval) - pr_err("could not set MSP sleepstate\n"); - } - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - writel(0, msp->registers + MSP_GCR); writel(0, msp->registers + MSP_TCF); writel(0, msp->registers + MSP_RCF); @@ -743,25 +710,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); msp->i2s_cont = i2s_cont; - msp->pinctrl_p = pinctrl_get(msp->dev); - if (IS_ERR(msp->pinctrl_p)) - dev_err(&pdev->dev, "could not get MSP pinctrl\n"); - else { - msp->pinctrl_def = pinctrl_lookup_state(msp->pinctrl_p, - PINCTRL_STATE_DEFAULT); - if (IS_ERR(msp->pinctrl_def)) { - dev_err(&pdev->dev, - "could not get MSP defstate (%li)\n", - PTR_ERR(msp->pinctrl_def)); - } - msp->pinctrl_sleep = pinctrl_lookup_state(msp->pinctrl_p, - PINCTRL_STATE_SLEEP); - if (IS_ERR(msp->pinctrl_sleep)) - dev_err(&pdev->dev, - "could not get MSP idlestate (%li)\n", - PTR_ERR(msp->pinctrl_def)); - } - return 0; } diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index e5cd105c90f9..8ce014eb37e8 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -528,12 +528,6 @@ struct ux500_msp { int loopback_enable; u32 backup_regs[MAX_MSP_BACKUP_REGS]; unsigned int f_bitclk; - /* Pin modes */ - struct pinctrl *pinctrl_p; - struct pinctrl_state *pinctrl_def; - struct pinctrl_state *pinctrl_sleep; - /* Reference Count */ - int pinctrl_rxtx_ref; }; struct ux500_msp_dma_params { -- cgit v1.2.1 From 7f92581b21707bfe09e14410283692b658b9ef10 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Fri, 24 May 2013 12:39:16 +0200 Subject: ASoC: ab8500-codec: Move codec ops on a separate structure Define ab8500 codec operations structure on its own rather than inline with snd_soc_dai_drivers to clean up the code and make the style coherent with other codec drivers. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 4ca45b9d9625..b8ba0adacfce 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2380,6 +2380,11 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } +static const struct snd_soc_dai_ops ab8500_codec_ops = { + .set_fmt = ab8500_codec_set_dai_fmt, + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, +}; + static struct snd_soc_dai_driver ab8500_codec_dai[] = { { .name = "ab8500-codec-dai.0", @@ -2391,12 +2396,7 @@ static struct snd_soc_dai_driver ab8500_codec_dai[] = { .rates = AB8500_SUPPORTED_RATE, .formats = AB8500_SUPPORTED_FMT, }, - .ops = (struct snd_soc_dai_ops[]) { - { - .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, - .set_fmt = ab8500_codec_set_dai_fmt, - } - }, + .ops = &ab8500_codec_ops, .symmetric_rates = 1 }, { @@ -2409,12 +2409,7 @@ static struct snd_soc_dai_driver ab8500_codec_dai[] = { .rates = AB8500_SUPPORTED_RATE, .formats = AB8500_SUPPORTED_FMT, }, - .ops = (struct snd_soc_dai_ops[]) { - { - .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, - .set_fmt = ab8500_codec_set_dai_fmt, - } - }, + .ops = &ab8500_codec_ops, .symmetric_rates = 1 } }; -- cgit v1.2.1 From b7230d7e4c1f6a87ddb96dbc106435e0dfee0f37 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Fri, 24 May 2013 12:39:17 +0200 Subject: ASoC: ux500: Drop dangling struct i2s_controller Drop struct i2s_controller from the ux500 ASoC driver as right now it is instantiated but not used anywhere. Also drop a mismatched device_unregister in the process. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.c | 19 ------------------- sound/soc/ux500/ux500_msp_i2s.h | 12 ------------ 2 files changed, 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index b029b2d673d7..cba0e86833e9 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -649,7 +649,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct msp_i2s_platform_data *platform_data) { struct resource *res = NULL; - struct i2s_controller *i2s_cont; struct device_node *np = pdev->dev.of_node; struct ux500_msp *msp; @@ -694,22 +693,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->msp_state = MSP_STATE_IDLE; msp->loopback_enable = 0; - /* I2S-controller is allocated and added in I2S controller class. */ - i2s_cont = devm_kzalloc(&pdev->dev, sizeof(*i2s_cont), GFP_KERNEL); - if (!i2s_cont) { - dev_err(&pdev->dev, - "%s: ERROR: Failed to allocate I2S-controller!\n", - __func__); - return -ENOMEM; - } - i2s_cont->dev.parent = &pdev->dev; - i2s_cont->data = (void *)msp; - i2s_cont->id = (s16)msp->id; - snprintf(i2s_cont->name, sizeof(i2s_cont->name), "ux500-msp-i2s.%04x", - msp->id); - dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); - msp->i2s_cont = i2s_cont; - return 0; } @@ -717,8 +700,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, struct ux500_msp *msp) { dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); - - device_unregister(&msp->i2s_cont->dev); } MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 8ce014eb37e8..ccfcc32b1c2b 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -469,17 +469,6 @@ struct i2s_message { size_t period_len; }; -struct i2s_controller { - struct module *owner; - unsigned int id; - unsigned int class; - const struct i2s_algorithm *algo; /* the algorithm to access the bus */ - void *data; - struct mutex bus_lock; - struct device dev; /* the controller device */ - char name[48]; -}; - struct ux500_msp_config { unsigned int f_inputclk; unsigned int rx_clk_sel; @@ -515,7 +504,6 @@ struct ux500_msp { enum enum_i2s_controller id; void __iomem *registers; struct device *dev; - struct i2s_controller *i2s_cont; struct stedma40_chan_cfg *dma_cfg_rx; struct stedma40_chan_cfg *dma_cfg_tx; struct dma_chan *tx_pipeid; -- cgit v1.2.1 From 2a357137fac4e2e92d13d37b161a6ff4535eecc6 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Fri, 24 May 2013 12:39:18 +0200 Subject: ASoC: ux500: Drop unused code from msp headers Drop unused fields and structures from ux500_msp_i2s header file, as those looks like leftover from a previous implementation of the driver. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.h | 31 ------------------------------- 2 files changed, 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index f53104359f15..c7212825fe4c 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -58,8 +58,6 @@ struct ux500_msp_i2s_drvdata { unsigned int rx_mask; int slots; int slot_width; - u8 configured; - int data_delay; /* Clocks */ unsigned int master_clk; diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index ccfcc32b1c2b..d5e41763c9c7 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -341,11 +341,6 @@ enum msp_compress_mode { MSP_COMPRESS_MODE_A_LAW = 3 }; -enum msp_spi_burst_mode { - MSP_SPI_BURST_MODE_DISABLE = 0, - MSP_SPI_BURST_MODE_ENABLE = 1 -}; - enum msp_expand_mode { MSP_EXPAND_MODE_LINEAR = 0, MSP_EXPAND_MODE_LINEAR_SIGNED = 1, @@ -454,21 +449,6 @@ struct msp_protdesc { u32 clocks_per_frame; }; -struct i2s_message { - enum i2s_direction_t i2s_direction; - void *txdata; - void *rxdata; - size_t txbytes; - size_t rxbytes; - int dma_flag; - int tx_offset; - int rx_offset; - bool cyclic_dma; - dma_addr_t buf_addr; - size_t buf_len; - size_t period_len; -}; - struct ux500_msp_config { unsigned int f_inputclk; unsigned int rx_clk_sel; @@ -480,8 +460,6 @@ struct ux500_msp_config { unsigned int tx_fsync_sel; unsigned int rx_fifo_config; unsigned int tx_fifo_config; - unsigned int spi_clk_mode; - unsigned int spi_burst_mode; unsigned int loopback_enable; unsigned int tx_data_enable; unsigned int default_protdesc; @@ -491,13 +469,9 @@ struct ux500_msp_config { unsigned int direction; unsigned int protocol; unsigned int frame_freq; - unsigned int frame_size; enum msp_data_size data_size; unsigned int def_elem_len; unsigned int iodelay; - void (*handler) (void *data); - void *tx_callback_data; - void *rx_callback_data; }; struct ux500_msp { @@ -506,15 +480,10 @@ struct ux500_msp { struct device *dev; struct stedma40_chan_cfg *dma_cfg_rx; struct stedma40_chan_cfg *dma_cfg_tx; - struct dma_chan *tx_pipeid; - struct dma_chan *rx_pipeid; enum msp_state msp_state; - int (*transfer) (struct ux500_msp *msp, struct i2s_message *message); - struct timer_list notify_timer; int def_elem_len; unsigned int dir_busy; int loopback_enable; - u32 backup_regs[MAX_MSP_BACKUP_REGS]; unsigned int f_bitclk; }; -- cgit v1.2.1 From 06b9671ee69d48a1436077805903e3d9c1ed9662 Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Fri, 24 May 2013 12:39:19 +0200 Subject: ASoC: ux500: Add missing mop500_ab8500.h include Add a missing include that was resulting in some sparse warning for non-static structure without forward declaration. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 884a36224fb1..5e0f14634271 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -24,6 +24,7 @@ #include "ux500_pcm.h" #include "ux500_msp_dai.h" +#include "mop500_ab8500.h" #include "../codecs/ab8500-codec.h" #define TX_SLOT_MONO 0x0008 -- cgit v1.2.1 From f82030f978ae21ee790a90610ff21ce72667958e Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Fri, 24 May 2013 12:39:20 +0200 Subject: ASoC: ux500: Drop redundant msp id enumerations Ux500 has two equivalent enum for device id, one in platform_data and one in a local header. Fix this by dropping the local one. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.h | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index d5e41763c9c7..189a3751993b 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -16,6 +16,7 @@ #define UX500_MSP_I2S_H #include +#include #define MSP_INPUT_FREQ_APB 48000000 @@ -365,13 +366,6 @@ enum msp_protocol { */ #define MAX_MSP_BACKUP_REGS 36 -enum enum_i2s_controller { - MSP_0_I2S_CONTROLLER = 0, - MSP_1_I2S_CONTROLLER, - MSP_2_I2S_CONTROLLER, - MSP_3_I2S_CONTROLLER, -}; - enum i2s_direction_t { MSP_DIR_TX = 0x01, MSP_DIR_RX = 0x02, @@ -475,7 +469,7 @@ struct ux500_msp_config { }; struct ux500_msp { - enum enum_i2s_controller id; + enum msp_i2s_id id; void __iomem *registers; struct device *dev; struct stedma40_chan_cfg *dma_cfg_rx; -- cgit v1.2.1 From 939d3c6a6c56d1db6ab7e44ddf11de60f0122d1a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 27 May 2013 09:46:53 +0100 Subject: ASoC: bells: Hookup DMICs for Bells Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index ceed466af9ff..29e246803626 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -350,8 +350,16 @@ static struct snd_soc_codec_conf bells_codec_conf[] = { }, }; +static struct snd_soc_dapm_widget bells_widgets[] = { + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + + { "DMIC", NULL, "MICBIAS2" }, + { "IN2L", NULL, "DMIC" }, + { "IN2R", NULL, "DMIC" }, }; static struct snd_soc_card bells_cards[] = { @@ -365,6 +373,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -383,6 +393,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -401,6 +413,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), -- cgit v1.2.1 From 4059a42c51f45566a7eb69f0c7af3ff65bee7a52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 May 2013 14:53:40 +0200 Subject: ALSA: hda - Enable mic-mute LED on more HP laptops The newer HP laptops have SSID 103c:20xx and 103c:21xx, and these usually have the mic-mute LED on Fn-F8. Let's enable it, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d6427e0bf..9b6cb270dbe5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2233,6 +2233,10 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Folio", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x1900, "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2000, + "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2100, + "HP", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389, -- cgit v1.2.1 From a130243b96622e16af7c1ac7ba903b4cec7aa81b Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Tue, 28 May 2013 16:16:39 +0200 Subject: ASoC: ux500: Ensure consistent configuration between DAIs Current implementation of mop500_ab8500 allows for inconsistent sample rate and channel count configuration between the playback and recording interfaces, through in the hardware the two MSP controllers share common clock and frame sync signals. This patch adds the necessary code to ensure that the two device are configure consistently. The check is added at machine driver level, as how to lock DAI configuration depend of the actual hardware implementation. Signed-off-by: Fabio Baltieri Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) (limited to 'sound') diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 5e0f14634271..7e923ecf8901 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include @@ -44,6 +45,12 @@ static unsigned int tx_slots = DEF_TX_SLOTS; static unsigned int rx_slots = DEF_RX_SLOTS; +/* Configuration consistency parameters */ +static DEFINE_MUTEX(mop500_ab8500_params_lock); +static unsigned long mop500_ab8500_usage; +static int mop500_ab8500_rate; +static int mop500_ab8500_channels; + /* Clocks */ static const char * const enum_mclk[] = { "SYSCLK", @@ -231,6 +238,21 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, substream->name, substream->number); + /* Ensure configuration consistency between DAIs */ + mutex_lock(&mop500_ab8500_params_lock); + if (mop500_ab8500_usage) { + if (mop500_ab8500_rate != params_rate(params) || + mop500_ab8500_channels != params_channels(params)) { + mutex_unlock(&mop500_ab8500_params_lock); + return -EBUSY; + } + } else { + mop500_ab8500_rate = params_rate(params); + mop500_ab8500_channels = params_channels(params); + } + __set_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + channels = params_channels(params); switch (params_format(params)) { @@ -329,9 +351,22 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, return 0; } +static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + mutex_lock(&mop500_ab8500_params_lock); + __clear_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + + return 0; +} + struct snd_soc_ops mop500_ab8500_ops[] = { { .hw_params = mop500_ab8500_hw_params, + .hw_free = mop500_ab8500_hw_free, .startup = mop500_ab8500_startup, .shutdown = mop500_ab8500_shutdown, } -- cgit v1.2.1 From 0c2e3f3420bb790a4e5bc14d3d50a722964ad73e Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 28 May 2013 12:01:50 +0100 Subject: ASoC: wm_adsp: Ensure set controls are synced on each boot Rename `dirty' to `set' as it is a bit more descriptive. A set control is any control that has been set by the user. We need to ensure that everytime we boot the DSP we sync out any controls that were set. We could at some point start keeping track of the default values of the controls to suppress some of the device I/O. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d715c8ede772..ddba3fea39eb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -242,7 +242,7 @@ struct wm_coeff_ctl { struct list_head list; void *cache; size_t len; - unsigned int dirty:1; + unsigned int set:1; struct snd_kcontrol *kcontrol; }; @@ -424,7 +424,7 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); if (!ctl->enabled) { - ctl->dirty = 1; + ctl->set = 1; return 0; } @@ -760,7 +760,7 @@ static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) list_for_each_entry(ctl, &wm_coeff->ctl_list, list) { - if (!ctl->enabled || ctl->dirty) + if (!ctl->enabled || ctl->set) continue; ret = wm_coeff_read_control(ctl->kcontrol, ctl->cache, @@ -781,13 +781,12 @@ static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) list) { if (!ctl->enabled) continue; - if (ctl->dirty) { + if (ctl->set) { ret = wm_coeff_write_control(ctl->kcontrol, ctl->cache, ctl->len); if (ret < 0) return ret; - ctl->dirty = 0; } } @@ -864,7 +863,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, goto err_ctl; } ctl->enabled = 1; - ctl->dirty = 0; + ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; ctl->card = codec->card->snd_card; @@ -1434,12 +1433,12 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - /* Initialize caches for enabled and non-dirty controls */ + /* Initialize caches for enabled and unset controls */ ret = wm_coeff_init_control_caches(dsp->wm_coeff); if (ret != 0) goto err; - /* Sync dirty controls */ + /* Sync set controls */ ret = wm_coeff_sync_controls(dsp->wm_coeff); if (ret != 0) goto err; @@ -1591,12 +1590,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - /* Initialize caches for enabled and non-dirty controls */ + /* Initialize caches for enabled and unset controls */ ret = wm_coeff_init_control_caches(dsp->wm_coeff); if (ret != 0) goto err; - /* Sync dirty controls */ + /* Sync set controls */ ret = wm_coeff_sync_controls(dsp->wm_coeff); if (ret != 0) goto err; -- cgit v1.2.1 From c375b2d7eff01d6423b95b2d44e8466beae0a15a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 28 May 2013 00:55:12 -0700 Subject: ASoC: fsi: fixup sparse errors This patch fixup below sparse errors ${LINUX}/sound/soc/sh/fsi.c:1459:9: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:1634:25: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:1639:17: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:2093:9: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:2105:9: \ error: incompatible types in conditional expression (different base types) Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index f830c41f97dd..30390260bb67 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -276,7 +276,7 @@ struct fsi_stream_handler { int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); - void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, + int (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, int enable); }; #define fsi_stream_handler_call(io, func, args...) \ @@ -1188,7 +1188,7 @@ static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io) samples); } -static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int enable) { struct fsi_master *master = fsi_get_master(fsi); @@ -1201,6 +1201,8 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io) @@ -1409,7 +1411,7 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int start) { struct fsi_master *master = fsi_get_master(fsi); @@ -1422,6 +1424,8 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) -- cgit v1.2.1 From 287d03e9cd41a5f60bf96f43f8efea454f1cf74e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 28 May 2013 12:52:07 +0100 Subject: ASoC: wm8994: Remove restore of DAC enable state It's not been needed since the regmap conversion. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f1c54af45dcf..a265fd42b700 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3131,22 +3131,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; int i, ret; - unsigned int val, mask; - - if (control->revision < 4) { - /* force a HW read */ - ret = regmap_read(control->regmap, - WM8994_POWER_MANAGEMENT_5, &val); - - /* modify the cache only */ - codec->cache_only = 1; - mask = WM8994_DAC1R_ENA | WM8994_DAC1L_ENA | - WM8994_DAC2R_ENA | WM8994_DAC2L_ENA; - val &= mask; - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, - mask, val); - codec->cache_only = 0; - } for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { if (!wm8994->fll_suspend[i].out) -- cgit v1.2.1 From f7dbd399efff631203be9f09c07f128df18a3ee4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 28 May 2013 12:52:09 +0100 Subject: ASoC: wm8994: Ensure lambda is zeroed for WM8994 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a265fd42b700..0805d6ff9ff7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2078,6 +2078,7 @@ static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, /* Move down to proper range now rounding is done */ fll->k = K / 10; + fll->lambda = 0; pr_debug("N=%x K=%x\n", fll->n, fll->k); break; -- cgit v1.2.1 From 20a24225d8f94fc56f74a9068684869d6deebea5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2013 12:37:32 +0200 Subject: ALSA: PCI: Remove superfluous pci_set_drvdata(pci, NULL) at remove As drvdata is cleared to NULL at probe failure or at removal by the driver core, we don't have to call pci_set_drvdata(pci, NULL) any longer in each driver. The only remaining pci_set_drvdata(NULL) is in azx_firmware_cb() in hda_intel.c. Since this function itself releases the card instance, we need to clear drvdata here as well, so that it won't be released doubly in the remove callback. Signed-off-by: Takashi Iwai --- sound/pci/ad1889.c | 1 - sound/pci/ali5451/ali5451.c | 1 - sound/pci/als300.c | 1 - sound/pci/als4000.c | 1 - sound/pci/asihpi/hpioctl.c | 1 - sound/pci/atiixp.c | 1 - sound/pci/atiixp_modem.c | 1 - sound/pci/au88x0/au88x0.c | 1 - sound/pci/aw2/aw2-alsa.c | 1 - sound/pci/azt3328.c | 1 - sound/pci/bt87x.c | 1 - sound/pci/ca0106/ca0106_main.c | 1 - sound/pci/cmipci.c | 1 - sound/pci/cs4281.c | 1 - sound/pci/cs46xx/cs46xx.c | 1 - sound/pci/cs5530.c | 1 - sound/pci/cs5535audio/cs5535audio.c | 1 - sound/pci/ctxfi/xfi.c | 1 - sound/pci/echoaudio/echoaudio.c | 1 - sound/pci/emu10k1/emu10k1.c | 1 - sound/pci/emu10k1/emu10k1x.c | 1 - sound/pci/ens1370.c | 1 - sound/pci/es1938.c | 1 - sound/pci/es1968.c | 1 - sound/pci/fm801.c | 1 - sound/pci/hda/hda_intel.c | 2 -- sound/pci/ice1712/ice1712.c | 1 - sound/pci/ice1712/ice1724.c | 1 - sound/pci/intel8x0.c | 1 - sound/pci/intel8x0m.c | 1 - sound/pci/korg1212/korg1212.c | 1 - sound/pci/lola/lola.c | 1 - sound/pci/lx6464es/lx6464es.c | 1 - sound/pci/maestro3.c | 1 - sound/pci/mixart/mixart.c | 1 - sound/pci/nm256/nm256.c | 1 - sound/pci/oxygen/oxygen_lib.c | 1 - sound/pci/pcxhr/pcxhr.c | 1 - sound/pci/riptide/riptide.c | 2 -- sound/pci/rme32.c | 1 - sound/pci/rme96.c | 1 - sound/pci/rme9652/hdsp.c | 1 - sound/pci/rme9652/hdspm.c | 1 - sound/pci/rme9652/rme9652.c | 1 - sound/pci/sis7019.c | 1 - sound/pci/sonicvibes.c | 1 - sound/pci/trident/trident.c | 1 - sound/pci/via82xx.c | 1 - sound/pci/via82xx_modem.c | 1 - sound/pci/vx222/vx222.c | 1 - sound/pci/ymfpci/ymfpci.c | 1 - 51 files changed, 53 deletions(-) (limited to 'sound') diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index ad8a31173939..d2b9d617aee5 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1046,7 +1046,6 @@ static void snd_ad1889_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 53754f5edeb1..3dfa12b670eb 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2298,7 +2298,6 @@ static int snd_ali_probe(struct pci_dev *pci, static void snd_ali_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver ali5451_driver = { diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 864c4310366b..591efb6eef05 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -282,7 +282,6 @@ static void snd_als300_remove(struct pci_dev *pci) { snd_als300_dbgcallenter(); snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); snd_als300_dbgcallleave(); } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 61efda2a4d94..ffc821b0139e 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -984,7 +984,6 @@ out: static void snd_card_als4000_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index ef5019fe5193..7f0272032fbb 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -445,7 +445,6 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev) if (pa->p_buffer) vfree(pa->p_buffer); - pci_set_drvdata(pci_dev, NULL); if (1) dev_info(&pci_dev->dev, "remove %04x:%04x,%04x:%04x,%04x, HPI index %d\n", diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 6e78c6789858..fe4c61bdb8ba 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1714,7 +1714,6 @@ static int snd_atiixp_probe(struct pci_dev *pci, static void snd_atiixp_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver atiixp_driver = { diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index d0bec7ba3b0d..cf29b9a1d65d 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1334,7 +1334,6 @@ static int snd_atiixp_probe(struct pci_dev *pci, static void snd_atiixp_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver atiixp_modem_driver = { diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index b157e1fadd8f..7059dd69e5e6 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -371,7 +371,6 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_vortex_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } // pci_driver definition diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 08e9a4702cbc..2925220d3fcf 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -392,7 +392,6 @@ static int snd_aw2_probe(struct pci_dev *pci, static void snd_aw2_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* open callback */ diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 1204a0fa3368..c8e121611593 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2725,7 +2725,6 @@ snd_azf3328_remove(struct pci_dev *pci) { snd_azf3328_dbgcallenter(); snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); snd_azf3328_dbgcallleave(); } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 9febe5509748..18802039497a 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -953,7 +953,6 @@ _error: static void snd_bt87x_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* default entries for all Bt87x cards - it's not exported */ diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 1610a5705970..f4db5587e86e 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1896,7 +1896,6 @@ static int snd_ca0106_probe(struct pci_dev *pci, static void snd_ca0106_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index c617435db6e6..2755ec5bcc25 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3317,7 +3317,6 @@ static int snd_cmipci_probe(struct pci_dev *pci, static void snd_cmipci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 6a8695069941..64659facd155 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1971,7 +1971,6 @@ static int snd_cs4281_probe(struct pci_dev *pci, static void snd_cs4281_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 6b0d8b50a305..b03498325d66 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -158,7 +158,6 @@ static int snd_card_cs46xx_probe(struct pci_dev *pci, static void snd_card_cs46xx_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver cs46xx_driver = { diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index dace827b45d1..c6b82c85e044 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -91,7 +91,6 @@ static int snd_cs5530_dev_free(struct snd_device *device) static void snd_cs5530_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static u8 snd_cs5530_mixer_read(unsigned long io, u8 reg) diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 7e4b13e2d12a..902bebd3b3fb 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -391,7 +391,6 @@ static void snd_cs5535audio_remove(struct pci_dev *pci) { olpc_quirks_cleanup(); snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver cs5535audio_driver = { diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index d01ffcb2b2f5..d464ad2fc7b7 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -122,7 +122,6 @@ error: static void ct_card_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 760cbff53210..05cfe551ce42 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2323,7 +2323,6 @@ static void snd_echo_remove(struct pci_dev *pci) chip = pci_get_drvdata(pci); if (chip) snd_card_free(chip->card); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 8c5010f7889c..9e1bd0c39a8c 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -202,7 +202,6 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, static void snd_card_emu10k1_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index cdff11d48ebd..56ad9d6f200d 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1623,7 +1623,6 @@ static int snd_emu10k1x_probe(struct pci_dev *pci, static void snd_emu10k1x_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } // PCI IDs diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index db2dc835171d..372f8ea91fca 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2497,7 +2497,6 @@ static int snd_audiopci_probe(struct pci_dev *pci, static void snd_audiopci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver ens137x_driver = { diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 8423403954ab..9213fb38921c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1881,7 +1881,6 @@ static int snd_es1938_probe(struct pci_dev *pci, static void snd_es1938_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver es1938_driver = { diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5ae0d1..714525154605 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2909,7 +2909,6 @@ static int snd_es1968_probe(struct pci_dev *pci, static void snd_es1968_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver es1968_driver = { diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 4f07fda5adf2..706c5b67b708 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1370,7 +1370,6 @@ static int snd_card_fm801_probe(struct pci_dev *pci, static void snd_card_fm801_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ac75975a4276..49dfad4a099e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3764,7 +3764,6 @@ static int azx_probe(struct pci_dev *pci, out_free: snd_card_free(card); - pci_set_drvdata(pci, NULL); return err; } @@ -3834,7 +3833,6 @@ static void azx_remove(struct pci_dev *pci) if (card) snd_card_free(card); - pci_set_drvdata(pci, NULL); } /* PCI IDs */ diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 806407a3973e..28ec872e54c0 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2807,7 +2807,6 @@ static void snd_ice1712_remove(struct pci_dev *pci) if (ice->card_info && ice->card_info->chip_exit) ice->card_info->chip_exit(ice); snd_card_free(card); - pci_set_drvdata(pci, NULL); } static struct pci_driver ice1712_driver = { diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ce70e7f113e0..500471778291 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2800,7 +2800,6 @@ static void snd_vt1724_remove(struct pci_dev *pci) if (ice->card_info && ice->card_info->chip_exit) ice->card_info->chip_exit(ice); snd_card_free(card); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index b8fe40531b9c..59c8aaebb91e 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -3364,7 +3364,6 @@ static int snd_intel8x0_probe(struct pci_dev *pci, static void snd_intel8x0_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver intel8x0_driver = { diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index fea09e8ea608..3573c1193665 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1328,7 +1328,6 @@ static int snd_intel8x0m_probe(struct pci_dev *pci, static void snd_intel8x0m_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver intel8x0m_driver = { diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 43b4228d9afe..9cf9829555d4 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2473,7 +2473,6 @@ snd_korg1212_probe(struct pci_dev *pci, static void snd_korg1212_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver korg1212_driver = { diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 322b638e8ec4..7307d97186cb 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -759,7 +759,6 @@ out_free: static void lola_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } /* PCI IDs */ diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 298bc9b72991..3230e57f246c 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1139,7 +1139,6 @@ out_free: static void snd_lx6464es_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index c76ac1411210..d5417360f51f 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2775,7 +2775,6 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_m3_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver m3_driver = { diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 934dec98e2ce..1e0f6ee193f0 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1377,7 +1377,6 @@ static int snd_mixart_probe(struct pci_dev *pci, static void snd_mixart_remove(struct pci_dev *pci) { snd_mixart_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver mixart_driver = { diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 6febedb05936..fe79fff4c6dc 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1746,7 +1746,6 @@ static int snd_nm256_probe(struct pci_dev *pci, static void snd_nm256_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9562dc63ba60..b0cb48adddc7 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -722,7 +722,6 @@ EXPORT_SYMBOL(oxygen_pci_probe); void oxygen_pci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } EXPORT_SYMBOL(oxygen_pci_remove); diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b97384ad946d..d379b284955b 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1691,7 +1691,6 @@ static int pcxhr_probe(struct pci_dev *pci, static void pcxhr_remove(struct pci_dev *pci) { pcxhr_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver pcxhr_driver = { diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 63c1c8041554..56cc891e395e 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2066,7 +2066,6 @@ static void snd_riptide_joystick_remove(struct pci_dev *pci) if (gameport) { release_region(gameport->io, 8); gameport_unregister_port(gameport); - pci_set_drvdata(pci, NULL); } } #endif @@ -2179,7 +2178,6 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_card_riptide_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver driver = { diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 0ecd4100713e..cc26346ae66b 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1981,7 +1981,6 @@ snd_rme32_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static void snd_rme32_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver rme32_driver = { diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 5fb88ac82aa9..2a8ad9d1a2ae 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -2390,7 +2390,6 @@ snd_rme96_probe(struct pci_dev *pci, static void snd_rme96_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver rme96_driver = { diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 94084cdb130c..4f255dfee450 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5412,7 +5412,6 @@ static int snd_hdsp_probe(struct pci_dev *pci, static void snd_hdsp_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver hdsp_driver = { diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9ea05e956474..ef3cbc044f0c 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6737,7 +6737,6 @@ static int snd_hdspm_probe(struct pci_dev *pci, static void snd_hdspm_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver hdspm_driver = { diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 773a67fff4cd..b96d9e1adf6d 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2628,7 +2628,6 @@ static int snd_rme9652_probe(struct pci_dev *pci, static void snd_rme9652_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver rme9652_driver = { diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index d59abe1682c5..f2639a0c5a65 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1481,7 +1481,6 @@ error_out: static void snd_sis7019_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver sis7019_driver = { diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index a2e7686e7ae3..2a46bf98af30 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1528,7 +1528,6 @@ static int snd_sonic_probe(struct pci_dev *pci, static void snd_sonic_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver sonicvibes_driver = { diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 1aefd6204a63..b3b588bc94c3 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -169,7 +169,6 @@ static int snd_trident_probe(struct pci_dev *pci, static void snd_trident_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver trident_driver = { diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index d756a3562706..3c511d0caf9e 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2646,7 +2646,6 @@ static int snd_via82xx_probe(struct pci_dev *pci, static void snd_via82xx_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver via82xx_driver = { diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 4f5fd80b7e56..ca190283cbd7 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1227,7 +1227,6 @@ static int snd_via82xx_probe(struct pci_dev *pci, static void snd_via82xx_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver via82xx_modem_driver = { diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index e2f1ab37e154..ab8a9b1bfb8e 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -254,7 +254,6 @@ static int snd_vx222_probe(struct pci_dev *pci, static void snd_vx222_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 01c49655a3c1..e8932b2e4a5d 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -347,7 +347,6 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, static void snd_card_ymfpci_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } static struct pci_driver ymfpci_driver = { -- cgit v1.2.1 From d2c69807156b8c162ce3f1e37b220d03d57af063 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2013 12:40:04 +0200 Subject: sound: OSS: Remove superfluous pci_set_dvdata(pci, NULL) Only kahlua.c has it. Signed-off-by: Takashi Iwai --- sound/oss/kahlua.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 2a44cc106459..12be1fb512dd 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -178,7 +178,6 @@ static int probe_one(struct pci_dev *pdev, const struct pci_device_id *ent) return 0; err_out_free: - pci_set_drvdata(pdev, NULL); kfree(hw_config); return 1; } @@ -187,7 +186,6 @@ static void remove_one(struct pci_dev *pdev) { struct address_info *hw_config = pci_get_drvdata(pdev); sb_dsp_unload(hw_config, 0); - pci_set_drvdata(pdev, NULL); kfree(hw_config); } -- cgit v1.2.1 From 8b5a1f9c46c2b78716794b8762edf659ec25a87d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2013 12:49:32 +0200 Subject: ALSA: ISA: Remove superfluous *_set_drvdata(NULL) calls Similarly like the previous commit for PCI drivers, remove dev_set_drvdata(NULL) and pnp_set_drvdata(NULL) calls in ISA drivers now. Signed-off-by: Takashi Iwai --- sound/isa/ad1848/ad1848.c | 1 - sound/isa/adlib.c | 1 - sound/isa/cmi8328.c | 1 - sound/isa/cmi8330.c | 1 - sound/isa/cs423x/cs4231.c | 1 - sound/isa/cs423x/cs4236.c | 2 -- sound/isa/es1688/es1688.c | 1 - sound/isa/es18xx.c | 2 -- sound/isa/galaxy/galaxy.c | 1 - sound/isa/gus/gusclassic.c | 1 - sound/isa/gus/gusextreme.c | 1 - sound/isa/gus/gusmax.c | 1 - sound/isa/gus/interwave.c | 1 - sound/isa/msnd/msnd_pinnacle.c | 1 - sound/isa/opl3sa2.c | 2 -- sound/isa/opti9xx/miro.c | 1 - sound/isa/opti9xx/opti92x-ad1848.c | 1 - sound/isa/sb/jazz16.c | 1 - sound/isa/sb/sb16.c | 1 - sound/isa/sb/sb8.c | 1 - sound/isa/sc6000.c | 1 - sound/isa/sscape.c | 1 - sound/isa/wavefront/wavefront.c | 1 - 23 files changed, 26 deletions(-) (limited to 'sound') diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index c214ecf45400..e3f455bd85cd 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -135,7 +135,6 @@ out: snd_card_free(card); static int snd_ad1848_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index d26545543732..35659218710f 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -101,7 +101,6 @@ out: snd_card_free(card); static int snd_adlib_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index a7369fe19a6f..f84f073fc1e8 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -418,7 +418,6 @@ static int snd_cmi8328_remove(struct device *pdev, unsigned int dev) snd_cmi8328_cfg_write(cmi->port, CFG2, 0); snd_cmi8328_cfg_write(cmi->port, CFG3, 0); snd_card_free(card); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index c707c52268ab..270b9659ef7f 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -651,7 +651,6 @@ static int snd_cmi8330_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index aa7a5d86e480..ba9a74eff3e0 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -151,7 +151,6 @@ out: snd_card_free(card); static int snd_cs4231_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 252e9fb37db3..69614acb2052 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -504,7 +504,6 @@ static int snd_cs423x_isa_remove(struct device *pdev, unsigned int dev) { snd_card_free(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } @@ -600,7 +599,6 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, static void snd_cs423x_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); - pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 102874a703d4..cdcfb57f1f0a 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -213,7 +213,6 @@ out: static int snd_es1688_isa_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 24380efe31a1..12978b864c3a 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2235,7 +2235,6 @@ static int snd_es18xx_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } @@ -2305,7 +2304,6 @@ static int snd_audiodrive_pnp_detect(struct pnp_dev *pdev, static void snd_audiodrive_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); - pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index 672184e3221a..81244e7cea5b 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -623,7 +623,6 @@ error: static int snd_galaxy_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 16bca4e96c08..1adc1b924f39 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -215,7 +215,6 @@ out: snd_card_free(card); static int snd_gusclassic_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 0b9c2426b49f..38e1e3260c24 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -344,7 +344,6 @@ out: snd_card_free(card); static int snd_gusextreme_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); - dev_set_drvdata(dev, NULL); return 0; } diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index c309a5d0e7e1..652d5d834620 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -357,7 +357,6 @@ static int snd_gusmax_probe(struct device *pdev, unsigned int dev) static int snd_gusmax_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 78bc5744e89a..9942691cc0ca 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -849,7 +849,6 @@ static int snd_interwave_isa_probe(struct device *pdev, static int snd_interwave_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index ddabb406b14c..81aeb934261a 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -1064,7 +1064,6 @@ cfg_error: static int snd_msnd_isa_remove(struct device *pdev, unsigned int dev) { snd_msnd_unload(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 075777a6cf0b..cc01c419b7e9 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -757,7 +757,6 @@ static int snd_opl3sa2_pnp_detect(struct pnp_dev *pdev, static void snd_opl3sa2_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); - pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM @@ -900,7 +899,6 @@ static int snd_opl3sa2_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c3da1df9371d..619753d96ca5 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1495,7 +1495,6 @@ static int snd_miro_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index b41ed8661b23..103b33373fd4 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -1035,7 +1035,6 @@ static int snd_opti9xx_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 4961da4e627c..356a6308392f 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -345,7 +345,6 @@ static int snd_jazz16_remove(struct device *devptr, unsigned int dev) { struct snd_card *card = dev_get_drvdata(devptr); - dev_set_drvdata(devptr, NULL); snd_card_free(card); return 0; } diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 50dbec454f98..a4130993955f 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -566,7 +566,6 @@ static int snd_sb16_isa_probe(struct device *pdev, unsigned int dev) static int snd_sb16_isa_remove(struct device *pdev, unsigned int dev) { snd_card_free(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 237d964ff8a6..a806ae90a944 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -208,7 +208,6 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) static int snd_sb8_remove(struct device *pdev, unsigned int dev) { snd_card_free(dev_get_drvdata(pdev)); - dev_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 5376ebff845e..09d481b3ba7f 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -698,7 +698,6 @@ static int snd_sc6000_remove(struct device *devptr, unsigned int dev) release_region(port[dev], 0x10); release_region(mss_port[dev], 4); - dev_set_drvdata(devptr, NULL); snd_card_free(card); return 0; } diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 42a009720b29..57b338973ede 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1200,7 +1200,6 @@ _release_card: static int snd_sscape_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index fe5dd982bd23..82dd76939fa0 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -581,7 +581,6 @@ static int snd_wavefront_isa_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); return 0; } -- cgit v1.2.1 From f35e839a3ce730063174caaab8bf63432be553cf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2013 12:50:59 +0200 Subject: ALSA: Remove the rest of *_set_drvdata(NULL) calls A few calls are still left in parport drivers after this commit, which I'm not quite sure yet. Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 2 -- sound/drivers/dummy.c | 1 - sound/parisc/harmony.c | 3 --- sound/sparc/dbri.c | 2 -- sound/spi/at73c213.c | 1 - 5 files changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index aa5d8034890b..1ca8dc2ccb89 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1076,8 +1076,6 @@ static int aaci_remove(struct amba_device *dev) { struct snd_card *card = amba_get_drvdata(dev); - amba_set_drvdata(dev, NULL); - if (card) { struct aaci *aaci = card->private_data; writel(0, aaci->base + AACI_MAINCR); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index fd798f753609..11048cc744d0 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1129,7 +1129,6 @@ static int snd_dummy_probe(struct platform_device *devptr) static int snd_dummy_remove(struct platform_device *devptr) { snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); return 0; } diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 0e66ba48d453..67f56a2cee6a 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -902,8 +902,6 @@ snd_harmony_free(struct snd_harmony *h) if (h->iobase) iounmap(h->iobase); - parisc_set_drvdata(h->dev, NULL); - kfree(h); return 0; } @@ -1016,7 +1014,6 @@ static int snd_harmony_remove(struct parisc_device *padev) { snd_card_free(parisc_get_drvdata(padev)); - parisc_set_drvdata(padev, NULL); return 0; } diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 75e6016d3efe..eee7afcae375 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2670,8 +2670,6 @@ static int dbri_remove(struct platform_device *op) snd_dbri_free(card->private_data); snd_card_free(card); - dev_set_drvdata(&op->dev, NULL); - return 0; } diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index a1a24b979ed2..8e3d9a6c7a3b 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1070,7 +1070,6 @@ out: ssc_free(chip->ssc); snd_card_free(card); - dev_set_drvdata(&spi->dev, NULL); return 0; } -- cgit v1.2.1 From 70bd3b298bbbd5a36c55af957bb3b5f727218918 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 May 2013 20:28:16 +0100 Subject: ASoC: wm8994: Defer declaration of open circuit microphones Provide a bit of debounce to handle pathological cases with slow input better by allowing the microphone detection to run for a bit longer. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 40 +++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 32 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0805d6ff9ff7..2c2a183da2b6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3513,6 +3513,31 @@ static void wm8958_button_det(struct snd_soc_codec *codec, u16 status) wm8994->btn_mask); } +static void wm8958_open_circuit_work(struct work_struct *work) +{ + struct wm8994_priv *wm8994 = container_of(work, + struct wm8994_priv, + open_circuit_work.work); + struct device *dev = wm8994->wm8994->dev; + + wm1811_micd_stop(wm8994->hubs.codec); + + mutex_lock(&wm8994->accdet_lock); + + dev_dbg(dev, "Reporting open circuit\n"); + + wm8994->jack_mic = false; + wm8994->mic_detecting = true; + + wm8958_micd_set_rate(wm8994->hubs.codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + wm8994->btn_mask | + SND_JACK_HEADSET); + + mutex_unlock(&wm8994->accdet_lock); +} + static void wm8958_mic_id(void *data, u16 status) { struct snd_soc_codec *codec = data; @@ -3522,16 +3547,9 @@ static void wm8958_mic_id(void *data, u16 status) if (!(status & WM8958_MICD_STS)) { /* If nothing present then clear our statuses */ dev_dbg(codec->dev, "Detected open circuit\n"); - wm8994->jack_mic = false; - wm8994->mic_detecting = true; - - wm1811_micd_stop(codec); - wm8958_micd_set_rate(codec); - - snd_soc_jack_report(wm8994->micdet[0].jack, 0, - wm8994->btn_mask | - SND_JACK_HEADSET); + schedule_delayed_work(&wm8994->open_circuit_work, + msecs_to_jiffies(2500)); return; } @@ -3812,6 +3830,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + cancel_delayed_work_sync(&wm8994->open_circuit_work); + pm_runtime_get_sync(codec->dev); /* We may occasionally read a detection without an impedence @@ -3911,6 +3931,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, wm1811_jackdet_bootstrap); + INIT_DELAYED_WORK(&wm8994->open_circuit_work, + wm8958_open_circuit_work); switch (control->type) { case WM8994: diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 55ddf4d57d9b..9d19a9185d35 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -134,6 +134,7 @@ struct wm8994_priv { struct mutex accdet_lock; struct wm8994_micdet micdet[2]; struct delayed_work mic_work; + struct delayed_work open_circuit_work; bool mic_detecting; bool jack_mic; int btn_mask; -- cgit v1.2.1 From e684533b1044498606a37e2b5ba8bb0bef56067f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 30 May 2013 10:06:01 +0100 Subject: ASoC: wm0010: Set IRQ as a wake source The DSPs IRQ should be a wake source as several of the possible algorithms may run whilst the AP is asleepi and require to wake the AP to push or pull more data, such as compressed playback. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 370af0cbcc9a..b6df319869ac 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -972,6 +972,13 @@ static int wm0010_spi_probe(struct spi_device *spi) } wm0010->irq = irq; + ret = irq_set_irq_wake(irq, 1); + if (ret) { + dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n", + irq, ret); + return ret; + } + if (spi->max_speed_hz) wm0010->board_max_spi_speed = spi->max_speed_hz; else -- cgit v1.2.1 From 18b494527bc3e6847e34d18e540c78a12c3aef2f Mon Sep 17 00:00:00 2001 From: Paul Bolle Date: Thu, 30 May 2013 09:37:40 +0200 Subject: ASoC: OMAP: Remove obsolete Makefile line Support for omap2evm was removed in v3.0. But only one of its two lines in this Makefile was removed. Remove the second line too. Signed-off-by: Paul Bolle Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/Makefile | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 2b225945359b..a725905b2c68 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -26,7 +26,6 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o -- cgit v1.2.1 From d65f63da22092176f3ffedcc013b801c7e7ba390 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:09 +0200 Subject: ASoC: blackfin: Remove unused bf5xx-{i2s, tdm, ac97}-pcm.h The structs defined in these files are completely unused, so remove both the structs and the files. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ad1836.c | 1 - sound/soc/blackfin/bf5xx-ad193x.c | 1 - sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 ------------------ 11 files changed, 78 deletions(-) delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 7e2f36004a5a..53f84085bf1f 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -39,7 +39,6 @@ #include -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h deleted file mode 100644 index d324d5826a9b..000000000000 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-ac97-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_AC97_PCM_H -#define _BF5XX_AC97_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d23f4b0ea54f..152817633256 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -30,7 +30,6 @@ #include "../codecs/ad1836.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad1836; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 0e55e9f2a514..ce773b31916f 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -39,7 +39,6 @@ #include "../codecs/ad193x.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad193x; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index b30f88bbd703..3450e8f9080d 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -48,7 +48,6 @@ #include "../codecs/ad1980.h" -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" static struct snd_soc_card bf5xx_board; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 61cc91d4a028..786bbdd96e7c 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -45,7 +45,6 @@ #include "../codecs/ad73311.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" #if CONFIG_SND_BF5XX_SPORT_NUM == 0 #define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 262c1de364d8..8726c3ad4947 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -39,7 +39,6 @@ #include -#include "bf5xx-i2s-pcm.h" #include "bf5xx-sport.h" static void bf5xx_dma_irq(void *data) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h deleted file mode 100644 index 0c2c5a68d4ff..000000000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-i2s-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_I2S_PCM_H -#define _BF5XX_I2S_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 7dbeef1099b4..9c19ccc936e2 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -40,7 +40,6 @@ #include #include "../codecs/ssm2602.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" static struct snd_soc_card bf5xx_ssm2602; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 19e855d5677e..a6b5457036ef 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -39,7 +39,6 @@ #include -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h deleted file mode 100644 index 7f8cc01c4477..000000000000 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2009 Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_TDM_PCM_H -#define _BF5XX_TDM_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -#endif -- cgit v1.2.1 From b7ede5dea0746611a75cf49cd3b2f64097c53ef5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:10 +0200 Subject: ASoC: blackfin: bf5xx-i2s: Use dev_{err, dbg} instead of pr_{error, debug} Using dev_{err,dbg} instead of pr_{error,debug} makes it easier to recognize which device created the message. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 24 +++++++++++++----------- 1 file changed, 13 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index dd0c2a4f83a3..78411f266f60 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -74,7 +74,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: - printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); + dev_err(cpu_dai->dev, "%s: Unknown DAI format type\n", + __func__); ret = -EINVAL; break; } @@ -88,7 +89,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: - printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); + dev_err(cpu_dai->dev, "%s: Unknown DAI master type\n", + __func__); ret = -EINVAL; break; } @@ -141,14 +143,14 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, bf5xx_i2s->rcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, bf5xx_i2s->tcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } } @@ -162,7 +164,7 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - pr_debug("%s enter\n", __func__); + dev_dbg(dai->dev, "%s enter\n", __func__); /* No active stream, SPORT is allowed to be configured again. */ if (!dai->active) bf5xx_i2s->configured = 0; @@ -173,7 +175,7 @@ static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - pr_debug("%s : sport %d\n", __func__, dai->id); + dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); if (dai->capture_active) sport_rx_stop(sport_handle); @@ -188,19 +190,19 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; int ret; - pr_debug("%s : sport %d\n", __func__, dai->id); + dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, bf5xx_i2s->rcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, bf5xx_i2s->tcr2, 0, 0); if (ret) { - pr_err("SPORT is busy!\n"); + dev_err(dai->dev, "SPORT is busy!\n"); return -EBUSY; } @@ -264,7 +266,7 @@ static int bf5xx_i2s_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &bf5xx_i2s_component, &bf5xx_i2s_dai, 1); if (ret) { - pr_err("Failed to register DAI: %d\n", ret); + dev_err(&pdev->dev, "Failed to register DAI: %d\n", ret); sport_done(sport_handle); return ret; } @@ -276,7 +278,7 @@ static int bf5xx_i2s_remove(struct platform_device *pdev) { struct sport_device *sport_handle = platform_get_drvdata(pdev); - pr_debug("%s enter\n", __func__); + dev_dbg(&pdev->dev, "%s enter\n", __func__); snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); -- cgit v1.2.1 From 634426048462373aba69c201390c3e75bc9d00d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:11 +0200 Subject: ASoC: blackfin: bf5xx-sport: Allow setting rx and tx mask independently Since the hardware supports it there is no need to artificially limit this to just being able to set the same mask for both tx and rx. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 8 ++++---- sound/soc/blackfin/bf5xx-sport.c | 10 +++++----- sound/soc/blackfin/bf5xx-sport.h | 2 +- sound/soc/blackfin/bf5xx-tdm.c | 4 ++-- 4 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 490217325975..c66bef826ac5 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -231,9 +231,9 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) return 0; #if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport, 16, 0x3FF, 1); + ret = sport_set_multichannel(sport, 16, 0x3FF, 0x3FF, 1); #else - ret = sport_set_multichannel(sport, 16, 0x1F, 1); + ret = sport_set_multichannel(sport, 16, 0x1F, 0x1F, 1); #endif if (ret) { pr_err("SPORT is busy!\n"); @@ -311,9 +311,9 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /*SPORT works in TDM mode to simulate AC97 transfers*/ #if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 0x3FF, 1); #else - ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); + ret = sport_set_multichannel(sport_handle, 16, 0x1F, 0x1F, 1); #endif if (ret) { pr_err("SPORT is busy!\n"); diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 2fd9f2a06968..695351241db8 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -46,10 +46,10 @@ /* note: multichannel is in units of 8 channels, * tdm_count is # channels NOT / 8 ! */ int sport_set_multichannel(struct sport_device *sport, - int tdm_count, u32 mask, int packed) + int tdm_count, u32 tx_mask, u32 rx_mask, int packed) { - pr_debug("%s tdm_count=%d mask:0x%08x packed=%d\n", __func__, - tdm_count, mask, packed); + pr_debug("%s tdm_count=%d tx_mask:0x%08x rx_mask:0x%08x packed=%d\n", + __func__, tdm_count, tx_mask, rx_mask, packed); if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) return -EBUSY; @@ -65,8 +65,8 @@ int sport_set_multichannel(struct sport_device *sport, sport->regs->mcmc2 = FRAME_DELAY | MCMEN | \ (packed ? (MCDTXPE|MCDRXPE) : 0); - sport->regs->mtcs0 = mask; - sport->regs->mrcs0 = mask; + sport->regs->mtcs0 = tx_mask; + sport->regs->mrcs0 = rx_mask; sport->regs->mtcs1 = 0; sport->regs->mrcs1 = 0; sport->regs->mtcs2 = 0; diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 5ab60bd613ea..9fc2192feb3b 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -128,7 +128,7 @@ void sport_done(struct sport_device *sport); /* note: multichannel is in units of 8 channels, tdm_count is number of channels * NOT / 8 ! all channels are enabled by default */ int sport_set_multichannel(struct sport_device *sport, int tdm_count, - u32 mask, int packed); + u32 tx_mask, u32 rx_mask, int packed); int sport_config_rx(struct sport_device *sport, unsigned int rcr1, unsigned int rcr2, diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 69e9a3e935bd..aa0851650b56 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -198,7 +198,7 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai) int ret; struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - ret = sport_set_multichannel(sport, 8, 0xFF, 1); + ret = sport_set_multichannel(sport, 8, 0xFF, 0xFF, 1); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; @@ -265,7 +265,7 @@ static int bfin_tdm_probe(struct platform_device *pdev) return -ENODEV; /* SPORT works in TDM mode */ - ret = sport_set_multichannel(sport_handle, 8, 0xFF, 1); + ret = sport_set_multichannel(sport_handle, 8, 0xFF, 0xFF, 1); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- cgit v1.2.1 From 569ef65a973e19ec3327c8efbcf26bfc844af7e3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:12 +0200 Subject: ASoC: blackfin: bf5xx-i2s: Allocate buffer only as large as requested There is no need to always allocate the maximum buffer size. While we are at it also pass errors returned by snd_pcm_lib_malloc_pages() on to the upper layers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 8726c3ad4947..107c1c9b1cb6 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -66,10 +66,7 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = { static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - snd_pcm_lib_malloc_pages(substream, size); - - return 0; + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) -- cgit v1.2.1 From a3935a29f68c261d31b41c896f95c9333b615abf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:13 +0200 Subject: ASoC: blackfin: bf5xx-i2s-pcm: Use snd_pcm_lib_preallocate_pages_for_all() Use snd_pcm_lib_preallocate_pages_for_all() for pre-allocating the DMA buffers instead of re-implementing the same functionality. Note that there is no need to call snd_pcm_lib_free_pages_for_all() since the ALSA core takes care of this for us. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 63 ++------------------------------------ 1 file changed, 3 insertions(+), 60 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 107c1c9b1cb6..9931a18c962e 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -209,55 +209,12 @@ static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .mmap = bf5xx_pcm_mmap, }; -static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) { - pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } - buf->bytes = size; - - pr_debug("%s, area:%p, size:0x%08lx\n", __func__, - buf->area, buf->bytes); - - return 0; -} - -static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(NULL, buf->bytes, buf->area, 0); - buf->area = NULL; - } -} - static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; pr_debug("%s enter\n", __func__); if (!card->dev->dma_mask) @@ -265,27 +222,13 @@ static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, card->dev, size, size); } static struct snd_soc_platform_driver bf5xx_i2s_soc_platform = { .ops = &bf5xx_pcm_i2s_ops, .pcm_new = bf5xx_pcm_i2s_new, - .pcm_free = bf5xx_pcm_free_dma_buffers, }; static int bfin_i2s_soc_platform_probe(struct platform_device *pdev) -- cgit v1.2.1 From 8b5e2e396b589119bcc9c6a382a999e0202bae18 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:14 +0200 Subject: ASoC: blackfin: bf5xx-i2s: Add support for TDM mode The bf5xx-i2s{,-pcm} and bf5xx-tdm{-pcm} drivers are nearly identical. Both are for the same hardware each supporting a slight different subset of the hardware. The bf5xx-i2s driver supports 2 channel I2S mode while the bf5xx-tdm driver supports TDM mode and channel remapping. This patch adds support for TDM mode and channel remapping to the bf5xx-i2s driver so that we'll eventually be able to retire the bf5xx-tdm driver. Unfortunately the hardware is fixed to using 8 channels in TDM mode. The bf5xx-tdm driver jumps through a few hoops to make it work well with other channel counts as well: * Don't support mmap * Translate between internal frame size (which is always 8 * sample_size) and ALSA frame size (which depends on the channel count) * Have special copy and silence callbacks which are aware of the mismatch between internal and ALSA frame size * Reduce the maximum buffer size to ensure that there is enough headroom for dummy data. The bf5xx-i2s driver is going to use the same mechanisms when being used int TDM mode. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 118 ++++++++++++++++++++++++++++++++++++- sound/soc/blackfin/bf5xx-i2s-pcm.h | 17 ++++++ sound/soc/blackfin/bf5xx-i2s.c | 105 ++++++++++++++++++++++++++++++--- 3 files changed, 229 insertions(+), 11 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 9931a18c962e..9cb4a80df98e 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -40,6 +40,7 @@ #include #include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" static void bf5xx_dma_irq(void *data) { @@ -49,7 +50,6 @@ static void bf5xx_dma_irq(void *data) static const struct snd_pcm_hardware bf5xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER, .formats = SNDRV_PCM_FMTBIT_S16_LE | @@ -66,7 +66,16 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = { static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int buffer_size = params_buffer_bytes(params); + struct bf5xx_i2s_pcm_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) + buffer_size = buffer_size / params_channels(params) * 8; + + return snd_pcm_lib_malloc_pages(substream, buffer_size); } static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) @@ -78,9 +87,16 @@ static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct sport_device *sport = runtime->private_data; int period_bytes = frames_to_bytes(runtime, runtime->period_size); + struct bf5xx_i2s_pcm_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) + period_bytes = period_bytes / runtime->channels * 8; pr_debug("%s enter\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -127,10 +143,15 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct sport_device *sport = runtime->private_data; unsigned int diff; snd_pcm_uframes_t frames; + struct bf5xx_i2s_pcm_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + pr_debug("%s enter\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff = sport_curr_offset_tx(sport); @@ -147,6 +168,8 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) diff = 0; frames = bytes_to_frames(substream->runtime, diff); + if (dma_data->tdm_mode) + frames = frames * runtime->channels / 8; return frames; } @@ -158,11 +181,18 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dma_buffer *buf = &substream->dma_buffer; + struct bf5xx_i2s_pcm_data *dma_data; int ret; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + pr_debug("%s enter\n", __func__); snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); + if (dma_data->tdm_mode) + runtime->hw.buffer_bytes_max /= 4; + else + runtime->hw.info |= SNDRV_PCM_INFO_MMAP; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -198,6 +228,88 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, return 0 ; } +static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int sample_size = runtime->sample_bits / 8; + struct bf5xx_i2s_pcm_data *dma_data; + unsigned int i; + void *src, *dst; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = buf; + dst = runtime->dma_area; + dst += pos * sample_size * 8; + + while (count--) { + for (i = 0; i < runtime->channels; i++) { + memcpy(dst + dma_data->map[i] * + sample_size, src, sample_size); + src += sample_size; + } + dst += 8 * sample_size; + } + } else { + src = runtime->dma_area; + src += pos * sample_size * 8; + dst = buf; + + while (count--) { + for (i = 0; i < runtime->channels; i++) { + memcpy(dst, src + dma_data->map[i] * + sample_size, sample_size); + dst += sample_size; + } + src += 8 * sample_size; + } + } + } else { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = buf; + dst = runtime->dma_area; + dst += frames_to_bytes(runtime, pos); + } else { + src = runtime->dma_area; + src += frames_to_bytes(runtime, pos); + dst = buf; + } + + memcpy(dst, src, frames_to_bytes(runtime, count)); + } + + return 0; +} + +static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int sample_size = runtime->sample_bits / 8; + void *buf = runtime->dma_area; + struct bf5xx_i2s_pcm_data *dma_data; + unsigned int offset, size; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (dma_data->tdm_mode) { + offset = pos * 8 * sample_size; + size = count * 8 * sample_size; + } else { + offset = frames_to_bytes(runtime, pos); + size = frames_to_bytes(runtime, count); + } + + snd_pcm_format_set_silence(runtime->format, buf + offset, size); + + return 0; +} + static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, @@ -207,6 +319,8 @@ static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .trigger = bf5xx_pcm_trigger, .pointer = bf5xx_pcm_pointer, .mmap = bf5xx_pcm_mmap, + .copy = bf5xx_pcm_copy, + .silence = bf5xx_pcm_silence, }; static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h new file mode 100644 index 000000000000..1f0435249f88 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.h @@ -0,0 +1,17 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _BF5XX_TDM_PCM_H +#define _BF5XX_TDM_PCM_H + +#define BFIN_TDM_DAI_MAX_SLOTS 8 + +struct bf5xx_i2s_pcm_data { + unsigned int map[BFIN_TDM_DAI_MAX_SLOTS]; + bool tdm_mode; +}; + +#endif diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 78411f266f60..9a174fc47d39 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -42,6 +42,7 @@ #include #include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" struct bf5xx_i2s_port { u16 tcr1; @@ -49,6 +50,13 @@ struct bf5xx_i2s_port { u16 tcr2; u16 rcr2; int configured; + + unsigned int slots; + unsigned int tx_mask; + unsigned int rx_mask; + + struct bf5xx_i2s_pcm_data tx_dma_data; + struct bf5xx_i2s_pcm_data rx_dma_data; }; static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -170,6 +178,64 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, bf5xx_i2s->configured = 0; } +static int bf5xx_i2s_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; + unsigned int tx_mapped = 0, rx_mapped = 0; + unsigned int slot; + int i; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_i2s->tx_dma_data.map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_i2s->rx_dma_data.map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + +static int bf5xx_i2s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; + + if (slots % 8 != 0 || slots > 8) + return -EINVAL; + + if (width != 32) + return -EINVAL; + + bf5xx_i2s->slots = slots; + bf5xx_i2s->tx_mask = tx_mask; + bf5xx_i2s->rx_mask = rx_mask; + + bf5xx_i2s->tx_dma_data.tdm_mode = slots != 0; + bf5xx_i2s->rx_dma_data.tdm_mode = slots != 0; + + return sport_set_multichannel(sport_handle, slots, tx_mask, rx_mask, 0); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { @@ -206,7 +272,8 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) return -EBUSY; } - return 0; + return sport_set_multichannel(sport_handle, bf5xx_i2s->slots, + bf5xx_i2s->tx_mask, bf5xx_i2s->rx_mask, 0); } #else @@ -214,6 +281,23 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) #define bf5xx_i2s_resume NULL #endif +static int bf5xx_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; + unsigned int i; + + for (i = 0; i < BFIN_TDM_DAI_MAX_SLOTS; i++) { + bf5xx_i2s->tx_dma_data.map[i] = i; + bf5xx_i2s->rx_dma_data.map[i] = i; + } + + dai->playback_dma_data = &bf5xx_i2s->tx_dma_data; + dai->capture_dma_data = &bf5xx_i2s->rx_dma_data; + + return 0; +} + #define BF5XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ @@ -226,22 +310,25 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, + .shutdown = bf5xx_i2s_shutdown, + .hw_params = bf5xx_i2s_hw_params, + .set_fmt = bf5xx_i2s_set_dai_fmt, + .set_tdm_slot = bf5xx_i2s_set_tdm_slot, + .set_channel_map = bf5xx_i2s_set_channel_map, }; static struct snd_soc_dai_driver bf5xx_i2s_dai = { + .probe = bf5xx_i2s_dai_probe, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .ops = &bf5xx_i2s_dai_ops, @@ -257,7 +344,7 @@ static int bf5xx_i2s_probe(struct platform_device *pdev) int ret; /* configure SPORT for I2S */ - sport_handle = sport_init(pdev, 4, 2 * sizeof(u32), + sport_handle = sport_init(pdev, 4, 8 * sizeof(u32), sizeof(struct bf5xx_i2s_port)); if (!sport_handle) return -ENODEV; -- cgit v1.2.1 From b88546324ef1b61fc6e844e56ad4e90169514fb7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:15 +0200 Subject: ASoC: blackfin: Switch bf5xx-ad193x from bf5xx-tdm to bf5xx-i2s The bf5xx-i2s driver now has support for TDM mode and the bf5xx-tdm driver is going to be removed soon, so switch the driver over to bf5xx-i2s. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 18 +++++++++--------- sound/soc/blackfin/bf5xx-ad193x.c | 39 ++++++++++----------------------------- 2 files changed, 19 insertions(+), 38 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 16b88f5c26e2..906c349a00ea 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -56,6 +56,15 @@ config SND_SOC_BFIN_EVAL_ADAV80X Note: This driver assumes that the ADAV80X digital record and playback interfaces are connected to the first SPORT port on the BF5XX board. +config SND_BF5XX_SOC_AD193X + tristate "SoC AD193X Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD193X + help + Say Y if you want to add support for AD193X codec on Blackfin. + This driver supports AD1936, AD1937, AD1938 and AD1939. + config SND_BF5XX_SOC_AD73311 tristate "SoC AD73311 Audio support for Blackfin" depends on SND_BF5XX_I2S @@ -90,15 +99,6 @@ config SND_BF5XX_SOC_AD1836 help Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. -config SND_BF5XX_SOC_AD193X - tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD193X - help - Say Y if you want to add support for AD193X codec on Blackfin. - This driver supports AD1936, AD1937, AD1938 and AD1939. - config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index ce773b31916f..603ad1f2b9b9 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -39,29 +39,16 @@ #include "../codecs/ad193x.h" -#include "bf5xx-tdm.h" - static struct snd_soc_card bf5xx_ad193x; -static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int bf5xx_ad193x_link_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int clk = 0; - unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; - int ret = 0; - - switch (params_rate(params)) { - case 48000: - clk = 24576000; - break; - } + int ret; /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -70,9 +57,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set cpu DAI channel mapping */ - ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), - channel_map, ARRAY_SIZE(channel_map), channel_map); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); if (ret < 0) return ret; @@ -82,30 +67,26 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, #define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) -static struct snd_soc_ops bf5xx_ad193x_ops = { - .hw_params = bf5xx_ad193x_hw_params, -}; - static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { { .name = "ad193x", .stream_name = "AD193X", - .cpu_dai_name = "bfin-tdm.0", + .cpu_dai_name = "bfin-i2s.0", .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-tdm-pcm-audio", + .platform_name = "bfin-i2s-pcm-audio", .codec_name = "spi0.5", - .ops = &bf5xx_ad193x_ops, .dai_fmt = BF5XX_AD193X_DAIFMT, + .init = bf5xx_ad193x_link_init, }, { .name = "ad193x", .stream_name = "AD193X", - .cpu_dai_name = "bfin-tdm.1", + .cpu_dai_name = "bfin-i2s.1", .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-tdm-pcm-audio", + .platform_name = "bfin-i2s-pcm-audio", .codec_name = "spi0.5", - .ops = &bf5xx_ad193x_ops, .dai_fmt = BF5XX_AD193X_DAIFMT, + .init = bf5xx_ad193x_link_init, }, }; -- cgit v1.2.1 From 34f4095564ff334adae5ab4a9904f8d66d03e994 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:16 +0200 Subject: ASoC: blackfin: Switch bf5xx-ad1836 from bf5xx-tdm to bf5xx-i2s The bf5xx-i2s driver now has support for TDM mode and the bf5xx-tdm driver is going to be removed soon, so switch the driver over to bf5xx-i2s. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 16 ++++++++-------- sound/soc/blackfin/bf5xx-ad1836.c | 18 +++++++----------- 2 files changed, 15 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 906c349a00ea..4a67865bc4fc 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -56,6 +56,14 @@ config SND_SOC_BFIN_EVAL_ADAV80X Note: This driver assumes that the ADAV80X digital record and playback interfaces are connected to the first SPORT port on the BF5XX board. +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" depends on SND_BF5XX_I2S @@ -91,14 +99,6 @@ config SND_BF5XX_TDM mode. You will also need to select the audio interfaces to support below. -config SND_BF5XX_SOC_AD1836 - tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1836 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 152817633256..8fcfc4ec3a51 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -30,14 +30,10 @@ #include "../codecs/ad1836.h" -#include "bf5xx-tdm.h" - static struct snd_soc_card bf5xx_ad1836; -static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int bf5xx_ad1836_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; @@ -48,13 +44,13 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + return 0; } -static struct snd_soc_ops bf5xx_ad1836_ops = { - .hw_params = bf5xx_ad1836_hw_params, -}; - #define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) @@ -62,9 +58,9 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai = { .name = "ad1836", .stream_name = "AD1836", .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .ops = &bf5xx_ad1836_ops, + .platform_name = "bfin-i2s-pcm-audio", .dai_fmt = BF5XX_AD1836_DAIFMT, + .init = bf5xx_ad1836_init, }; static struct snd_soc_card bf5xx_ad1836 = { -- cgit v1.2.1 From cc37961b21eb3d57d421ca34ffec9bbe0a6096c0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 28 May 2013 19:22:17 +0200 Subject: ASoC: blackfin: Remove bf5xx-tdm driver Now that the bf5xx-i2s driver supports TDM mode and all users of the bf5xx-tdm driver have been switch over to using the bf5xx-i2s driver there is no need to keep the b5fxx-tdm driver around. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 13 -- sound/soc/blackfin/Makefile | 4 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 343 ------------------------------------- sound/soc/blackfin/bf5xx-tdm.c | 328 ----------------------------------- sound/soc/blackfin/bf5xx-tdm.h | 23 --- 5 files changed, 711 deletions(-) delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm.h (limited to 'sound') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 4a67865bc4fc..54f74f8cbb75 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -89,16 +89,6 @@ config SND_BFIN_AD73311_SE Enter the GPIO used to control AD73311's SE pin. Acceptable values are 0 to 7 -config SND_BF5XX_TDM - tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" - depends on (BLACKFIN && SND_SOC) - select SND_BF5XX_SOC_SPORT - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in TDM - mode. - You will also need to select the audio interfaces to support below. - config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN @@ -174,9 +164,6 @@ config SND_BF5XX_SOC_I2S config SND_BF6XX_SOC_I2S tristate -config SND_BF5XX_SOC_TDM - tristate - config SND_BF5XX_SOC_AC97 tristate diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 6fea1f4cbee2..ad0a6e99bc5d 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -1,23 +1,19 @@ # Blackfin Platform Support snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o -snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o -snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o -obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o -obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support snd-ad1836-objs := bf5xx-ad1836.o diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c deleted file mode 100644 index a6b5457036ef..000000000000 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ /dev/null @@ -1,343 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-tdm-pcm.c - * Author: Barry Song - * - * Created: Tue June 06 2009 - * Description: DMA driver for tdm codec - * - * Modified: - * Copyright 2009 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include - -#include "bf5xx-tdm.h" -#include "bf5xx-sport.h" - -#define PCM_BUFFER_MAX 0x8000 -#define FRAGMENT_SIZE_MIN (4*1024) -#define FRAGMENTS_MIN 2 -#define FRAGMENTS_MAX 32 - -static void bf5xx_dma_irq(void *data) -{ - struct snd_pcm_substream *pcm = data; - snd_pcm_period_elapsed(pcm); -} - -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_48000, - .channels_min = 2, - .channels_max = 8, - .buffer_bytes_max = PCM_BUFFER_MAX, - .period_bytes_min = FRAGMENT_SIZE_MIN, - .period_bytes_max = PCM_BUFFER_MAX/2, - .periods_min = FRAGMENTS_MIN, - .periods_max = FRAGMENTS_MAX, -}; - -static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - snd_pcm_lib_malloc_pages(substream, size * 4); - - return 0; -} - -static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - - return 0; -} - -static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int fragsize_bytes = frames_to_bytes(runtime, runtime->period_size); - - fragsize_bytes /= runtime->channels; - /* inflate the fragsize to match the dma width of SPORT */ - fragsize_bytes *= 8; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, runtime->dma_area, - runtime->periods, fragsize_bytes); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, runtime->dma_area, - runtime->periods, fragsize_bytes); - } - - return 0; -} - -static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_start(sport); - else - sport_rx_start(sport); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_stop(sport); - else - sport_rx_stop(sport); - break; - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int diff; - snd_pcm_uframes_t frames; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - diff = sport_curr_offset_tx(sport); - frames = diff / (8*4); /* 32 bytes per frame */ - } else { - diff = sport_curr_offset_rx(sport); - frames = diff / (8*4); - } - return frames; -} - -static int bf5xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - if (sport_handle != NULL) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - - runtime->private_data = sport_handle; - } else { - pr_err("sport_handle is NULL\n"); - ret = -ENODEV; - } -out: - return ret; -} - -static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - struct bf5xx_tdm_port *tdm_port = sport->private_data; - unsigned int *src; - unsigned int *dst; - int i; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = buf; - dst = (unsigned int *)substream->runtime->dma_area; - - dst += pos * 8; - while (count--) { - for (i = 0; i < substream->runtime->channels; i++) - *(dst + tdm_port->tx_map[i]) = *src++; - dst += 8; - } - } else { - src = (unsigned int *)substream->runtime->dma_area; - dst = buf; - - src += pos * 8; - while (count--) { - for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src + tdm_port->rx_map[i]); - src += 8; - } - } - - return 0; -} - -static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, - int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) -{ - unsigned char *buf = substream->runtime->dma_area; - buf += pos * 8 * 4; - memset(buf, '\0', count * 8 * 4); - - return 0; -} - -static struct snd_pcm_ops bf5xx_pcm_tdm_ops = { - .open = bf5xx_pcm_open, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = bf5xx_pcm_hw_params, - .hw_free = bf5xx_pcm_hw_free, - .prepare = bf5xx_pcm_prepare, - .trigger = bf5xx_pcm_trigger, - .pointer = bf5xx_pcm_pointer, - .copy = bf5xx_pcm_copy, - .silence = bf5xx_pcm_silence, -}; - -static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size * 4, - &buf->addr, GFP_KERNEL); - if (!buf->area) { - pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } - buf->bytes = size; - - return 0; -} - -static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(NULL, buf->bytes, buf->area, 0); - buf->area = NULL; - } -} - -static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); - -static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &bf5xx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } -out: - return ret; -} - -static struct snd_soc_platform_driver bf5xx_tdm_soc_platform = { - .ops = &bf5xx_pcm_tdm_ops, - .pcm_new = bf5xx_pcm_tdm_new, - .pcm_free = bf5xx_pcm_free_dma_buffers, -}; - -static int bf5xx_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &bf5xx_tdm_soc_platform); -} - -static int bf5xx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver bfin_tdm_driver = { - .driver = { - .name = "bfin-tdm-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = bf5xx_soc_platform_probe, - .remove = bf5xx_soc_platform_remove, -}; - -module_platform_driver(bfin_tdm_driver); - -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c deleted file mode 100644 index aa0851650b56..000000000000 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ /dev/null @@ -1,328 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-tdm.c - * Author: Barry Song - * - * Created: Thurs June 04 2009 - * Description: Blackfin I2S(TDM) CPU DAI driver - * Even though TDM mode can be as part of I2S DAI, but there - * are so much difference in configuration and data flow, - * it's very ugly to integrate I2S and TDM into a module - * - * Modified: - * Copyright 2009 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include "bf5xx-sport.h" -#include "bf5xx-tdm.h" - -static int bf5xx_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - int ret = 0; - - /* interface format:support TDM,slave mode */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - break; - default: - printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); - ret = -EINVAL; - break; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - break; - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFM: - ret = -EINVAL; - break; - default: - printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); - ret = -EINVAL; - break; - } - - return ret; -} - -static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; - int ret = 0; - - bf5xx_tdm->tcr2 &= ~0x1f; - bf5xx_tdm->rcr2 &= ~0x1f; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S32_LE: - bf5xx_tdm->tcr2 |= 31; - bf5xx_tdm->rcr2 |= 31; - sport_handle->wdsize = 4; - break; - /* at present, we only support 32bit transfer */ - default: - pr_err("not supported PCM format yet\n"); - return -EINVAL; - break; - } - - if (!bf5xx_tdm->configured) { - /* - * TX and RX are not independent,they are enabled at the - * same time, even if only one side is running. So, we - * need to configure both of them at the time when the first - * stream is opened. - * - * CPU DAI:slave mode. - */ - ret = sport_config_rx(sport_handle, bf5xx_tdm->rcr1, - bf5xx_tdm->rcr2, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport_handle, bf5xx_tdm->tcr1, - bf5xx_tdm->tcr2, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - bf5xx_tdm->configured = 1; - } - - return 0; -} - -static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; - - /* No active stream, SPORT is allowed to be configured again. */ - if (!dai->active) - bf5xx_tdm->configured = 0; -} - -static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; - int i; - unsigned int slot; - unsigned int tx_mapped = 0, rx_mapped = 0; - - if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || - (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) - return -EINVAL; - - for (i = 0; i < tx_num; i++) { - slot = tx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(tx_mapped & (1 << slot)))) { - bf5xx_tdm->tx_map[i] = slot; - tx_mapped |= 1 << slot; - } else - return -EINVAL; - } - for (i = 0; i < rx_num; i++) { - slot = rx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(rx_mapped & (1 << slot)))) { - bf5xx_tdm->rx_map[i] = slot; - rx_mapped |= 1 << slot; - } else - return -EINVAL; - } - - return 0; -} - -#ifdef CONFIG_PM -static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - if (dai->playback_active) - sport_tx_stop(sport); - if (dai->capture_active) - sport_rx_stop(sport); - - /* isolate sync/clock pins from codec while sports resume */ - peripheral_free_list(sport->pin_req); - - return 0; -} - -static int bf5xx_tdm_resume(struct snd_soc_dai *dai) -{ - int ret; - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - ret = sport_set_multichannel(sport, 8, 0xFF, 0xFF, 1); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - } - - ret = sport_config_rx(sport, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - } - - ret = sport_config_tx(sport, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - } - - peripheral_request_list(sport->pin_req, "soc-audio"); - - return 0; -} - -#else -#define bf5xx_tdm_suspend NULL -#define bf5xx_tdm_resume NULL -#endif - -static const struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { - .hw_params = bf5xx_tdm_hw_params, - .set_fmt = bf5xx_tdm_set_dai_fmt, - .shutdown = bf5xx_tdm_shutdown, - .set_channel_map = bf5xx_tdm_set_channel_map, -}; - -static struct snd_soc_dai_driver bf5xx_tdm_dai = { - .suspend = bf5xx_tdm_suspend, - .resume = bf5xx_tdm_resume, - .playback = { - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, - .ops = &bf5xx_tdm_dai_ops, -}; - -static const struct snd_soc_component_driver bf5xx_tdm_component = { - .name = "bf5xx-tdm", -}; - -static int bfin_tdm_probe(struct platform_device *pdev) -{ - struct sport_device *sport_handle; - int ret; - - /* configure SPORT for TDM */ - sport_handle = sport_init(pdev, 4, 8 * sizeof(u32), - sizeof(struct bf5xx_tdm_port)); - if (!sport_handle) - return -ENODEV; - - /* SPORT works in TDM mode */ - ret = sport_set_multichannel(sport_handle, 8, 0xFF, 0xFF, 1); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_rx(sport_handle, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_tx(sport_handle, 0, 0x1F, 0, 0); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = snd_soc_register_component(&pdev->dev, &bf5xx_tdm_component, - &bf5xx_tdm_dai, 1); - if (ret) { - pr_err("Failed to register DAI: %d\n", ret); - goto sport_config_err; - } - - return 0; - -sport_config_err: - sport_done(sport_handle); - return ret; -} - -static int bfin_tdm_remove(struct platform_device *pdev) -{ - struct sport_device *sport_handle = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - sport_done(sport_handle); - - return 0; -} - -static struct platform_driver bfin_tdm_driver = { - .probe = bfin_tdm_probe, - .remove = bfin_tdm_remove, - .driver = { - .name = "bfin-tdm", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(bfin_tdm_driver); - -/* Module information */ -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("TDM driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h deleted file mode 100644 index e986a3ea3315..000000000000 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ /dev/null @@ -1,23 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-tdm.h - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_TDM_H -#define _BF5XX_TDM_H - -#define BFIN_TDM_DAI_MAX_SLOTS 8 -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; - unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; - int configured; -}; - -#endif -- cgit v1.2.1 From 50941968fc9e359a89da2136b11328fe700dbd7d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 May 2013 13:42:26 +0100 Subject: ASoC: wm8994: Add digital loopback paths There is loopback control within the audio interfaces, provide control of this as there are some obscure scenarios where this could be used in production. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2c2a183da2b6..55a5cc639b90 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1494,6 +1494,24 @@ static const char *aif1dac_text[] = { "AIF1DACDAT", "AIF3DACDAT", }; +static const char *loopback_text[] = { + "None", "ADCDAT", +}; + +static const struct soc_enum aif1_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif1_loopback = + SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); + +static const struct soc_enum aif2_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif2_loopback = + SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); + static const struct soc_enum aif1dac_enum = SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); @@ -1740,6 +1758,9 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_MUX("AIF1 Loopback", SND_SOC_NOPM, 0, 0, &aif1_loopback), +SND_SOC_DAPM_MUX("AIF2 Loopback", SND_SOC_NOPM, 0, 0, &aif2_loopback), + SND_SOC_DAPM_POST("Debug log", post_ev), }; @@ -1871,9 +1892,9 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, - { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1 Loopback" }, { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, - { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2 Loopback" }, { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, @@ -1924,6 +1945,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + /* Loopback */ + { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, + { "AIF1 Loopback", "None", "AIF1DACDAT" }, + { "AIF2 Loopback", "ADCDAT", "AIF2ADCDAT" }, + { "AIF2 Loopback", "None", "AIF2DACDAT" }, + /* Sidetone */ { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, { "Left Sidetone", "DMIC2", "DMIC2L" }, -- cgit v1.2.1 From 8e0d70434d497f0265ccfe5d92a6a509410685ba Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Thu, 30 May 2013 16:13:09 +0200 Subject: ALSA: es1968: Add radio support for MediaForte M56VAP Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card. The GPIO connection type is automatically detected (like snd-fm801 driver). Also add a safety subsystem vendor check to skip radio detection if vendor differs from ESS (so we don't touch GPIOs on laptops). Tested with SF64-PCE2 and M56VAP cards. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 73 +++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 50 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 714525154605..5e2ec9687731 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; -- cgit v1.2.1 From a8cd7148045bd6a14adb15985dda806d17e9cab2 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 31 May 2013 12:57:09 +0200 Subject: ALSA: hdspm - Add support for 128-192kHz WordClock input Allow WordClock input rates of 128, 176.4 and 192kHz. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ef3cbc044f0c..e070ea85c30a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -400,8 +400,8 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */ #define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */ -#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */ -/* missing Bit for 111=128, 1000=176.4, 1001=192 */ +#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, 111=128 */ +#define HDSPM_wc_freq3 0x800 /* 1000=176.4, 1001=192 */ #define HDSPM_SyncRef0 0x10000 /* Sync Reference */ #define HDSPM_SyncRef1 0x20000 @@ -412,13 +412,17 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync) -#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2|\ + HDSPM_wc_freq3) #define HDSPM_wcFreq32 (HDSPM_wc_freq0) #define HDSPM_wcFreq44_1 (HDSPM_wc_freq1) #define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1) #define HDSPM_wcFreq64 (HDSPM_wc_freq2) #define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2) #define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq128 (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq176_4 (HDSPM_wc_freq3) +#define HDSPM_wcFreq192 (HDSPM_wc_freq0|HDSPM_wc_freq3) #define HDSPM_status1_F_0 0x0400000 #define HDSPM_status1_F_1 0x0800000 @@ -1181,6 +1185,15 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) case HDSPM_wcFreq96: rate = 96000; break; + case HDSPM_wcFreq128: + rate = 128000; + break; + case HDSPM_wcFreq176_4: + rate = 176400; + break; + case HDSPM_wcFreq192: + rate = 192000; + break; default: rate = 0; break; -- cgit v1.2.1 From a8a729fa06164889da4cacaecebe48370329716b Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 31 May 2013 12:57:10 +0200 Subject: ALSA: hdspm - Refactor SS/DS/QS clock multiplier into function When the DoubleSpeed or QuadSpeed bit is set, the SingleSpeed frequency has to be multiplied accordingly. Since this functionality will be required at least twice, refactor it into a separate function. The second reference to the newly introduced hdspm_rate_multiplier() will be in a separate commit. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 37 ++++++++++++++++++++++--------------- 1 file changed, 22 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e070ea85c30a..8eb2070fd258 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1091,6 +1091,26 @@ static int hdspm_round_frequency(int rate) return 48000; } +/* QS and DS rates normally can not be detected + * automatically by the card. Only exception is MADI + * in 96k frame mode. + * + * So if we read SS values (32 .. 48k), check for + * user-provided DS/QS bits in the control register + * and multiply the base frequency accordingly. + */ +static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) +{ + if (rate <= 48000) { + if (hdspm->control_register & HDSPM_QuadSpeed) + return rate * 4; + else if (hdspm->control_register & + HDSPM_DoubleSpeed) + return rate * 2; + }; + return rate; +} + static int hdspm_tco_sync_check(struct hdspm *hdspm); static int hdspm_sync_in_sync_check(struct hdspm *hdspm); @@ -1268,21 +1288,8 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) } } - /* QS and DS rates normally can not be detected - * automatically by the card. Only exception is MADI - * in 96k frame mode. - * - * So if we read SS values (32 .. 48k), check for - * user-provided DS/QS bits in the control register - * and multiply the base frequency accordingly. - */ - if (rate <= 48000) { - if (hdspm->control_register & HDSPM_QuadSpeed) - rate *= 4; - else if (hdspm->control_register & - HDSPM_DoubleSpeed) - rate *= 2; - } + rate = hdspm_rate_multiplier(hdspm, rate); + break; } -- cgit v1.2.1 From 7b5593976c2cae886afb920885580e300ebb01ca Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 31 May 2013 12:57:11 +0200 Subject: ALSA: hdspm - Allow SingleSpeed WordClock when in DS/QS mode Similarly to MADI, WordClock can also be at SingleSpeed while the card is actually working at twice or four times this rate. If so, multiply the base rate accordingly. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8eb2070fd258..bd501931ee23 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1225,7 +1225,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) */ if (rate != 0 && (status2 & HDSPM_SelSyncRefMask) == HDSPM_SelSyncRef_WORD) - return rate; + return hdspm_rate_multiplier(hdspm, rate); /* maybe a madi input (which is taken if sel sync is madi) */ if (status & HDSPM_madiLock) { -- cgit v1.2.1 From 2d01e39b9073361317eb72b390dc2a4a3d76e192 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 May 2013 13:42:30 +0100 Subject: ASoC: wm8994: Remove unused variable Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 55a5cc639b90..5d046b1c5a7e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3157,7 +3157,6 @@ static int wm8994_codec_suspend(struct snd_soc_codec *codec) static int wm8994_codec_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { -- cgit v1.2.1 From 2da1c4bf765cb32024e5db6fa75dab92916fa3b1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 May 2013 13:42:29 +0100 Subject: ASoC: wm8994: Allow debounce before MICDET identification For systems which do not have a jack detection feature allow some debounce to be specified before we perform accessory identification, improving robustness without impacting button detection responsiveness. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 35 +++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8994.h | 2 ++ 2 files changed, 35 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5d046b1c5a7e..0f58f003dc1a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3660,6 +3660,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) pm_runtime_get_sync(codec->dev); + cancel_delayed_work_sync(&wm8994->mic_complete_work); + mutex_lock(&wm8994->accdet_lock); reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); @@ -3842,11 +3844,33 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(wm8958_mic_detect); +static void wm8958_mic_work(struct work_struct *work) +{ + struct wm8994_priv *wm8994 = container_of(work, + struct wm8994_priv, + mic_complete_work.work); + struct snd_soc_codec *codec = wm8994->hubs.codec; + + dev_crit(codec->dev, "MIC WORK %x\n", wm8994->mic_status); + + pm_runtime_get_sync(codec->dev); + + mutex_lock(&wm8994->accdet_lock); + + wm8994->mic_id_cb(wm8994->mic_id_cb_data, wm8994->mic_status); + + mutex_unlock(&wm8994->accdet_lock); + + pm_runtime_put(codec->dev); + + dev_crit(codec->dev, "MIC WORK %x DONE\n", wm8994->mic_status); +} + static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->hubs.codec; - int reg, count, ret; + int reg, count, ret, id_delay; /* * Jack detection may have detected a removal simulataneously @@ -3856,6 +3880,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + cancel_delayed_work_sync(&wm8994->mic_complete_work); cancel_delayed_work_sync(&wm8994->open_circuit_work); pm_runtime_get_sync(codec->dev); @@ -3904,8 +3929,12 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) } } + wm8994->mic_status = reg; + id_delay = wm8994->wm8994->pdata.mic_id_delay; + if (wm8994->mic_detecting) - wm8994->mic_id_cb(wm8994->mic_id_cb_data, reg); + schedule_delayed_work(&wm8994->mic_complete_work, + msecs_to_jiffies(id_delay)); else wm8958_button_det(codec, reg); @@ -3971,6 +4000,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + INIT_DELAYED_WORK(&wm8994->mic_complete_work, wm8958_mic_work); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 9d19a9185d35..6536f8d45ac6 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -135,6 +135,8 @@ struct wm8994_priv { struct wm8994_micdet micdet[2]; struct delayed_work mic_work; struct delayed_work open_circuit_work; + struct delayed_work mic_complete_work; + u16 mic_status; bool mic_detecting; bool jack_mic; int btn_mask; -- cgit v1.2.1 From fd8b96574456e23fe7dad491711f371f86034c64 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 3 Jun 2013 15:57:55 +0100 Subject: ASoC: wm0010: Clear IRQ as wake source and include missing header Both clear the IRQ as being a wake source when we are finished with it and include a missing header file that is required. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index b6df319869ac..f5e835662cdc 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -14,6 +14,7 @@ #include #include +#include #include #include #include @@ -1002,6 +1003,8 @@ static int wm0010_spi_remove(struct spi_device *spi) gpio_set_value_cansleep(wm0010->gpio_reset, wm0010->gpio_reset_value); + irq_set_irq_wake(wm0010->irq, 0); + if (wm0010->irq) free_irq(wm0010->irq, wm0010); -- cgit v1.2.1 From ea421eb18d89935211bea80ba36e239e016e6e54 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Wed, 22 May 2013 16:53:37 +0530 Subject: ASoC: davinci: Remove redundant use of of_match_ptr macro 'mcasp_dt_ids' is always compiled in. Hence of_match_ptr is not necessary. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ecfc72f2e9..94f1e46ca79b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1023,7 +1023,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct device_node *np = pdev->dev.of_node; struct snd_platform_data *pdata = NULL; const struct of_device_id *match = - of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev); + of_match_device(mcasp_dt_ids, &pdev->dev); const u32 *of_serial_dir32; u8 *of_serial_dir; @@ -1256,7 +1256,7 @@ static struct platform_driver davinci_mcasp_driver = { .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, - .of_match_table = of_match_ptr(mcasp_dt_ids), + .of_match_table = mcasp_dt_ids, }, }; -- cgit v1.2.1 From 2f41a3f48a1400234a4213851453dcb4274322f5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 3 Jun 2013 11:37:41 -0600 Subject: ASoC: tegra: implement suspend/resume for Tegra30 AHUB Add tegra30_ahub_{suspend,resume}. These use regcache functions to restore all HW registers after power loss during a suspend/resume cycle. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_ahub.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 23e592f453fa..0f4787c24e43 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -627,9 +627,30 @@ static int tegra30_ahub_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int tegra30_ahub_suspend(struct device *dev) +{ + regcache_mark_dirty(ahub->regmap_ahub); + regcache_mark_dirty(ahub->regmap_apbif); + + return 0; +} + +static int tegra30_ahub_resume(struct device *dev) +{ + int ret; + + ret = regcache_sync(ahub->regmap_ahub); + ret |= regcache_sync(ahub->regmap_apbif); + + return ret; +} +#endif + static const struct dev_pm_ops tegra30_ahub_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, tegra30_ahub_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(tegra30_ahub_suspend, tegra30_ahub_resume) }; static struct platform_driver tegra30_ahub_driver = { -- cgit v1.2.1 From 5c5b08286fa4d782e44cae8738cf4328a29c4326 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 3 Jun 2013 11:37:42 -0600 Subject: ASoC: tegra: implement suspend/resume for Tegra30 I2S Add tegra30_i2s_{suspend,resume}. These use regcache functions to restore all HW registers after power loss during a suspend/resume cycle. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_i2s.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 31d092d83c71..bd0ebc09c8be 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -514,6 +514,24 @@ static int tegra30_i2s_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int tegra30_i2s_suspend(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + + regcache_mark_dirty(i2s->regmap); + + return 0; +} + +static int tegra30_i2s_resume(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + + return regcache_sync(i2s->regmap); +} +#endif + static const struct of_device_id tegra30_i2s_of_match[] = { { .compatible = "nvidia,tegra30-i2s", }, {}, @@ -522,6 +540,7 @@ static const struct of_device_id tegra30_i2s_of_match[] = { static const struct dev_pm_ops tegra30_i2s_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, tegra30_i2s_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(tegra30_i2s_suspend, tegra30_i2s_resume) }; static struct platform_driver tegra30_i2s_driver = { -- cgit v1.2.1 From dd4d2d6dfb49e8916064f2cb07f0ad7b32a82fb7 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Tue, 28 May 2013 20:55:56 +0200 Subject: ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order The VAG_POWER must be enabled after all other bits in CHIP_ANA_POWER and disabled before any other bit in CHIP_ANA_POWER. See the SGTL5000 datasheet (Table 31, BIT 7, page 42-43). Failing to follow this order will result in ugly loud "POP" noise at the end of playback. To achieve such order, use the _PRE and _POST DAPM widgets to trigger the power_vag_event, where the event type check has to be fixed accordingly as well. Signed-off-by: Marek Vasut Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20fc7ec3..b4297416401e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -153,12 +153,12 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -219,12 +219,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -232,16 +231,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- cgit v1.2.1 From 249e66c32679a24706ec182256a79bf7b1dac9a2 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 4 Jun 2013 12:58:14 -0600 Subject: ASoC: tegra: add runtime PM to resume functions Tegra HW needs clocks etc. active when touching registers. Make sure they are active during resume, by calling pm_runtime_get_sync() before touching HW. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_ahub.c | 4 ++++ sound/soc/tegra/tegra30_i2s.c | 9 ++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 0f4787c24e43..d554d46d08b5 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -640,8 +640,12 @@ static int tegra30_ahub_resume(struct device *dev) { int ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; ret = regcache_sync(ahub->regmap_ahub); ret |= regcache_sync(ahub->regmap_apbif); + pm_runtime_put(dev); return ret; } diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index bd0ebc09c8be..d04146cad61f 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -527,8 +527,15 @@ static int tegra30_i2s_suspend(struct device *dev) static int tegra30_i2s_resume(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; - return regcache_sync(i2s->regmap); + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(i2s->regmap); + pm_runtime_put(dev); + + return ret; } #endif -- cgit v1.2.1 From 02b504d9d8e565081b99176e7e464821d76fc994 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Mon, 3 Jun 2013 11:53:10 +0200 Subject: ALSA: hda - add mic fixup for ALC269VB on Ordissimo EVE2 This fixes the internal and external mic on Ordissimo EVE2, also known as Malata PC-B1303. We still don't know how to detect mic jack like Realtek's windows driver. Signed-off-by: Anisse Astier Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d2e91a9ab6..4c2ced178c82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3225,6 +3225,7 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269VB_FIXUP_ORDISSIMO_EVE2, }; static const struct hda_fixup alc269_fixups[] = { @@ -3467,6 +3468,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, }, + [ALC269VB_FIXUP_ORDISSIMO_EVE2] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x18, 0x03a11d20 }, /* mic */ + { 0x19, 0x411111f0 }, /* Unused bogus pin */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3533,6 +3543,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 /* Below is a quirk table taken from the old code. -- cgit v1.2.1 From 2c38d990fbdfc76176b03d60bc5e1a93d270760d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 5 Jun 2013 15:25:18 +0200 Subject: ALSA: hda/via - Use standard snd_hda_shutup_pins() Just a minor clean up. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e5245544eb52..cf31b664d2ed 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -480,14 +480,9 @@ static int via_suspend(struct hda_codec *codec) struct via_spec *spec = codec->spec; vt1708_stop_hp_work(codec); - if (spec->codec_type == VT1802) { - /* Fix pop noise on headphones */ - int i; - for (i = 0; i < spec->gen.autocfg.hp_outs; i++) - snd_hda_codec_write(codec, spec->gen.autocfg.hp_pins[i], - 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - 0x00); - } + /* Fix pop noise on headphones */ + if (spec->codec_type == VT1802) + snd_hda_shutup_pins(codec); return 0; } -- cgit v1.2.1 From 6dc6a3f81ee66b50acee5f1aa1de2f7d2d4e55fa Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Jun 2013 14:51:31 +0100 Subject: ASoC: arizona: Hookup SYSCLK to inputs and noise generators All sources and sinks should enable SYSCLK. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 11 +++++++++++ sound/soc/codecs/wm5110.c | 13 +++++++++++++ 2 files changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 9a186ff22b55..842adfdcb2a9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1388,10 +1388,21 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "OUT5L", NULL, "SYSCLK" }, { "OUT5R", NULL, "SYSCLK" }, + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + { "IN3L", NULL, "SYSCLK" }, + { "IN3R", NULL, "SYSCLK" }, + { "MICBIAS1", NULL, "MICVDD" }, { "MICBIAS2", NULL, "MICVDD" }, { "MICBIAS3", NULL, "MICVDD" }, + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + { "Noise Generator", NULL, "NOISE" }, { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index f53062f8c42e..17fdd9dbb17a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -820,10 +820,23 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "OUT6L", NULL, "SYSCLK" }, { "OUT6R", NULL, "SYSCLK" }, + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + { "IN3L", NULL, "SYSCLK" }, + { "IN3R", NULL, "SYSCLK" }, + { "IN4L", NULL, "SYSCLK" }, + { "IN4R", NULL, "SYSCLK" }, + { "MICBIAS1", NULL, "MICVDD" }, { "MICBIAS2", NULL, "MICVDD" }, { "MICBIAS3", NULL, "MICVDD" }, + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + { "Noise Generator", NULL, "NOISE" }, { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, -- cgit v1.2.1 From bf7c6e6ccbde22c96c5c1e5cec08740c31229df1 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Thu, 16 May 2013 14:08:07 +0800 Subject: ASoC: omap-hdmi-codec: make the driver common for other SoCs to support HDMI on CSR SiRFprimaII and atlasVI, we need one more HDMI pseudo codec, rather than add a new driver, we can make omap HDMI codec common for other SoCs as well. then the omap-hdmi codec becomes a generic HDMI pseudo- codec as HDMI audio features depend on HDMI specification not on SoCs. Signed-off-by: Barry Song Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 +-- sound/soc/codecs/Makefile | 4 +-- sound/soc/codecs/hdmi.c | 69 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/omap-hdmi.c | 69 ----------------------------------------- sound/soc/omap/omap-hdmi-card.c | 2 +- 5 files changed, 74 insertions(+), 74 deletions(-) create mode 100644 sound/soc/codecs/hdmi.c delete mode 100644 sound/soc/codecs/omap-hdmi.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00e31b0..d8c4f3dcf4a1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C - select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI + select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -287,7 +287,7 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_OMAP_HDMI_CODEC +config SND_SOC_HDMI_CODEC tristate config SND_SOC_PCM3008 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9a1f4c..49ff12718bed 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -41,7 +41,7 @@ snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o -snd-soc-omap-hdmi-codec-objs := omap-hdmi.o +snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -168,7 +168,7 @@ obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o -obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c new file mode 100644 index 000000000000..2bcae2b40c92 --- /dev/null +++ b/sound/soc/codecs/hdmi.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC codec driver for HDMI audio codecs. + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Ricardo Neri + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ +#include +#include + +#define DRV_NAME "hdmi-audio-codec" + +static struct snd_soc_codec_driver hdmi_codec; + +static struct snd_soc_dai_driver hdmi_codec_dai = { + .name = "hdmi-hifi", + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static int hdmi_codec_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &hdmi_codec, + &hdmi_codec_dai, 1); +} + +static int hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver hdmi_codec_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, +}; + +module_platform_driver(hdmi_codec_driver); + +MODULE_AUTHOR("Ricardo Neri "); +MODULE_DESCRIPTION("ASoC generic HDMI codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/omap-hdmi.c deleted file mode 100644 index 529d06444c54..000000000000 --- a/sound/soc/codecs/omap-hdmi.c +++ /dev/null @@ -1,69 +0,0 @@ -/* - * ALSA SoC codec driver for HDMI audio on OMAP processors. - * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ - * Author: Ricardo Neri - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ -#include -#include - -#define DRV_NAME "hdmi-audio-codec" - -static struct snd_soc_codec_driver omap_hdmi_codec; - -static struct snd_soc_dai_driver omap_hdmi_codec_dai = { - .name = "omap-hdmi-hifi", - .playback = { - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE, - }, -}; - -static int omap_hdmi_codec_probe(struct platform_device *pdev) -{ - return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, - &omap_hdmi_codec_dai, 1); -} - -static int omap_hdmi_codec_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); - return 0; -} - -static struct platform_driver omap_hdmi_codec_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, - - .probe = omap_hdmi_codec_probe, - .remove = omap_hdmi_codec_remove, -}; - -module_platform_driver(omap_hdmi_codec_driver); - -MODULE_AUTHOR("Ricardo Neri "); -MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c index d4eaa92e518e..7e66e9cba5a8 100644 --- a/sound/soc/omap/omap-hdmi-card.c +++ b/sound/soc/omap/omap-hdmi-card.c @@ -35,7 +35,7 @@ static struct snd_soc_dai_link omap_hdmi_dai = { .cpu_dai_name = "omap-hdmi-audio-dai", .platform_name = "omap-pcm-audio", .codec_name = "hdmi-audio-codec", - .codec_dai_name = "omap-hdmi-hifi", + .codec_dai_name = "hdmi-hifi", }; static struct snd_soc_card snd_soc_omap_hdmi = { -- cgit v1.2.1 From 2bdc1bb2b4e1f517d8aa5bbbad9cb6ccac8a94fb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 3 Jun 2013 10:20:39 +0100 Subject: ASoC: sgtl5000: Make device cache only when powered off When the regulators have been disabled mark the device as cache only so that we don't try to interact with the hardware. Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b4297416401e..c8f2afb74706 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -907,10 +907,25 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); + + regcache_cache_only(sgtl5000->regmap, false); + + ret = regcache_sync(sgtl5000->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + + regcache_cache_only(sgtl5000->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + return ret; + } } break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl5000->regmap, true); regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); break; -- cgit v1.2.1 From 3722dc8ebf041aedb1075078bd6728a38539a2aa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 5 Jun 2013 19:33:03 +0100 Subject: ASoC: max98090: Guard runtime PM callbacks Otherwise the functions will be defined but unreferenced when runtime PM is disabled, generating warnings. Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index cbb272b1f73d..3854d2209676 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2350,6 +2350,7 @@ static int max98090_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2369,6 +2370,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } +#endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, -- cgit v1.2.1 From 36bb00d4b2463b0b8b37fb0ce753813bf729b997 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Jun 2013 12:10:24 +0200 Subject: ALSA: hda - Drop hard dependency on CONFIG_SND_DYNAMIC_MINORS Currently HDMI codec driver sets the hard dependency (reverse selection) on CONFIG_SND_DYNAMIC_MINORS because the recent codecs may support more than two PCMs. But, this doesn't mean that we need always this option, since there can be a single PCM stream even with the modern codecs. This patch drops the hard dependency again but give more sensible error message when no enough PCMs are available due to the lack of this option. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_codec.c | 19 +++++++++++++++---- 2 files changed, 15 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 80a7d44bcf81..0c5371abecd2 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -140,7 +140,6 @@ config SND_HDA_CODEC_VIA config SND_HDA_CODEC_HDMI bool "Build HDMI/DisplayPort HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include HDMI and DisplayPort HD-audio codec diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 55108b5fb291..679fba44bdb5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4461,12 +4461,13 @@ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { /* * get the empty PCM device number to assign - * - * note the max device number is limited by HDA_MAX_PCMS, currently 10 */ -static int get_empty_pcm_device(struct hda_bus *bus, int type) +static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) { /* audio device indices; not linear to keep compatibility */ + /* assigned to static slots up to dev#10; if more needed, assign + * the later slot dynamically (when CONFIG_SND_DYNAMIC_MINORS=y) + */ static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, @@ -4480,18 +4481,28 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) return -EINVAL; } - for (i = 0; audio_idx[type][i] >= 0 ; i++) + for (i = 0; audio_idx[type][i] >= 0; i++) { +#ifndef CONFIG_SND_DYNAMIC_MINORS + if (audio_idx[type][i] >= 8) + break; +#endif if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; + } +#ifdef CONFIG_SND_DYNAMIC_MINORS /* non-fixed slots starting from 10 */ for (i = 10; i < 32; i++) { if (!test_and_set_bit(i, bus->pcm_dev_bits)) return i; } +#endif snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); +#ifndef CONFIG_SND_DYNAMIC_MINORS + snd_printk(KERN_WARNING "Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y\n"); +#endif return -EAGAIN; } -- cgit v1.2.1 From e7ecc27e520a9c9891362c6dabd18c4da9885946 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Jun 2013 14:00:23 +0200 Subject: ALSA: hda - Introduce bit flags to snd_hda_codec_read/write() snd_hda_codec_read(), snd_hda_codec_write() & co take the argument "direct" that indicates whether the given NID is a direct reference or an indirect reference. However, the indirect reference is practically unimplemented and never exists. And moreover, we don't need the indication of indirect reference at this high level, as NID can be represented in 16bit values at this point. Meanwhile, there are some cases where it'd be nice to give some operational options to these functions. So, we can reuse this argument as a new bit flag! Pretty frugal, eh? All callers so far pass zero to this argument, thus there is no behavior change by this replacement. The real usage of this new bit option will be added in the following patches. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++--------------------- sound/pci/hda/hda_codec.h | 8 ++++---- 2 files changed, 24 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 679fba44bdb5..503869aad7f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -185,20 +185,19 @@ EXPORT_SYMBOL_HDA(snd_hda_get_jack_type); * Compose a 32bit command word to be sent to the HD-audio controller */ static inline unsigned int -make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, +make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { u32 val; - if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + if ((codec->addr & ~0xf) || (nid & ~0x7f) || (verb & ~0xfff) || (parm & ~0xffff)) { - printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", - codec->addr, direct, nid, verb, parm); + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x\n", + codec->addr, nid, verb, parm); return ~0; } val = (u32)codec->addr << 28; - val |= (u32)direct << 27; val |= (u32)nid << 20; val |= verb << 8; val |= parm; @@ -209,7 +208,7 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, * Send and receive a verb */ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, - unsigned int *res) + int flags, unsigned int *res) { struct hda_bus *bus = codec->bus; int err; @@ -255,7 +254,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -264,12 +263,12 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * Returns the obtained response value, or -1 for an error. */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm) { - unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - if (codec_exec_verb(codec, cmd, &res)) + if (codec_exec_verb(codec, cmd, flags, &res)) return -1; return res; } @@ -279,7 +278,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -287,12 +286,12 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm) +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) { - unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned int cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - return codec_exec_verb(codec, cmd, + return codec_exec_verb(codec, cmd, flags, codec->bus->sync_write ? &res : NULL); } EXPORT_SYMBOL_HDA(snd_hda_codec_write); @@ -3582,7 +3581,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * snd_hda_codec_write_cache - send a single command with caching * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3591,7 +3590,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { int err; struct hda_cache_head *c; @@ -3600,7 +3599,7 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, cache_only = codec->cached_write; if (!cache_only) { - err = snd_hda_codec_write(codec, nid, direct, verb, parm); + err = snd_hda_codec_write(codec, nid, flags, verb, parm); if (err < 0) return err; } @@ -3624,7 +3623,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * snd_hda_codec_update_cache - check cache and write the cmd only when needed * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3635,7 +3634,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { struct hda_cache_head *c; u32 key; @@ -3651,7 +3650,7 @@ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, return 0; } mutex_unlock(&codec->bus->cmd_mutex); - return snd_hda_codec_write_cache(codec, nid, direct, verb, parm); + return snd_hda_codec_write_cache(codec, nid, flags, verb, parm); } EXPORT_SYMBOL_HDA(snd_hda_codec_update_cache); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f9021f452..39a658e02988 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -945,9 +945,9 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec); * low level functions */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm); -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); #define snd_hda_param_read(codec, nid, param) \ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) @@ -986,11 +986,11 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); /* cached write */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_sequence_write_cache(struct hda_codec *codec, const struct hda_verb *seq); int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_codec_resume_cache(struct hda_codec *codec); /* both for cmd & amp caches */ void snd_hda_codec_flush_cache(struct hda_codec *codec); -- cgit v1.2.1 From 63e51fd708f511a5989da04c669647993bc1a512 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Jun 2013 14:20:19 +0200 Subject: ALSA: hda - Don't take unresponsive D3 transition too serious When a codec is powered off, some systems don't respond properly after D3 FG transition, while the driver still expects the response and tries to fall back to different modes (polling and single-cmd). When the fallback happens, the driver stays in that mode, and falling back to the single-cmd mode means it'll loose the unsol event handling, too. The unresponsiveness at D3 isn't too serious, thus this fallback is mostly superfluous. We can gracefully ignore the error there so that the driver keeps the normal operation mode. This patch adds a new bit flag for codec read/write, set in the power transition stage, which is notified to the controller driver via a new bus->no_response_fallback flag. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 7 ++++++- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_intel.c | 3 +++ 3 files changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 503869aad7f9..35090b3acbac 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -221,6 +221,8 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, again: snd_hda_power_up(codec); mutex_lock(&bus->cmd_mutex); + if (flags & HDA_RW_NO_RESPONSE_FALLBACK) + bus->no_response_fallback = 1; for (;;) { trace_hda_send_cmd(codec, cmd); err = bus->ops.command(bus, cmd); @@ -233,6 +235,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, *res = bus->ops.get_response(bus, codec->addr); trace_hda_get_response(codec, *res); } + bus->no_response_fallback = 0; mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); if (!codec_in_pm(codec) && res && *res == -1 && bus->rirb_error) { @@ -3805,11 +3808,13 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, hda_nid_t fg = codec->afg ? codec->afg : codec->mfg; int count; unsigned int state; + int flags = 0; /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { /* transition time less than 10ms for power down */ msleep(codec->epss ? 10 : 100); + flags = HDA_RW_NO_RESPONSE_FALLBACK; } /* repeat power states setting at most 10 times*/ @@ -3818,7 +3823,7 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, codec->patch_ops.set_power_state(codec, fg, power_state); else { - snd_hda_codec_read(codec, fg, 0, + snd_hda_codec_read(codec, fg, flags, AC_VERB_SET_POWER_STATE, power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 39a658e02988..701c2e069b10 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -679,6 +679,7 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ + unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ }; @@ -930,6 +931,8 @@ enum { HDA_INPUT, HDA_OUTPUT }; +/* snd_hda_codec_read/write optional flags */ +#define HDA_RW_NO_RESPONSE_FALLBACK (1 << 0) /* * constructors diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 49dfad4a099e..f089fa0aa03d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -942,6 +942,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } + if (!bus->no_response_fallback) + return -1; + if (!chip->polling_mode && chip->poll_count < 2) { snd_printdd(SFX "%s: azx_get_response timeout, " "polling the codec once: last cmd=0x%08x\n", -- cgit v1.2.1 From e75a52c6723a61a0d768ee53794e86b7edbe54f0 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 6 Jun 2013 19:38:45 +0800 Subject: ASoC: WM8962: Create default platform data structure Embed a copy of struct wm8962_pdata in stuct wm8962_priv so that there's no need to check validity of pdata any more. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 68 +++++++++++++++++++++++++---------------------- 1 file changed, 36 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e9710280e5e1..d56dd867057d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -51,6 +51,7 @@ static const char *wm8962_supply_names[WM8962_NUM_SUPPLIES] = { /* codec private data */ struct wm8962_priv { + struct wm8962_pdata pdata; struct regmap *regmap; struct snd_soc_codec *codec; @@ -2345,12 +2346,13 @@ static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { static int wm8962_add_widgets(struct snd_soc_codec *codec) { - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + struct wm8962_pdata *pdata = &wm8962->pdata; struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_add_codec_controls(codec, wm8962_snd_controls, ARRAY_SIZE(wm8962_snd_controls)); - if (pdata && pdata->spk_mono) + if (pdata->spk_mono) snd_soc_add_codec_controls(codec, wm8962_spk_mono_controls, ARRAY_SIZE(wm8962_spk_mono_controls)); else @@ -2360,7 +2362,7 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8962_dapm_widgets, ARRAY_SIZE(wm8962_dapm_widgets)); - if (pdata && pdata->spk_mono) + if (pdata->spk_mono) snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_mono_widgets, ARRAY_SIZE(wm8962_dapm_spk_mono_widgets)); else @@ -2369,7 +2371,7 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8962_intercon, ARRAY_SIZE(wm8962_intercon)); - if (pdata && pdata->spk_mono) + if (pdata->spk_mono) snd_soc_dapm_add_routes(dapm, wm8962_spk_mono_intercon, ARRAY_SIZE(wm8962_spk_mono_intercon)); else @@ -3333,14 +3335,14 @@ static struct gpio_chip wm8962_template_chip = { static void wm8962_init_gpio(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int ret; wm8962->gpio_chip = wm8962_template_chip; wm8962->gpio_chip.ngpio = WM8962_MAX_GPIO; wm8962->gpio_chip.dev = codec->dev; - if (pdata && pdata->gpio_base) + if (pdata->gpio_base) wm8962->gpio_chip.base = pdata->gpio_base; else wm8962->gpio_chip.base = -1; @@ -3373,7 +3375,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; u16 *reg_cache = codec->reg_cache; int i, trigger, irq_pol; bool dmicclk, dmicdat; @@ -3421,30 +3423,28 @@ static int wm8962_probe(struct snd_soc_codec *codec) WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, 0); - if (pdata) { - /* Apply static configuration for GPIOs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) - if (pdata->gpio_init[i]) { - wm8962_set_gpio_mode(codec, i + 1); - snd_soc_write(codec, 0x200 + i, - pdata->gpio_init[i] & 0xffff); - } + /* Apply static configuration for GPIOs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) + if (pdata->gpio_init[i]) { + wm8962_set_gpio_mode(codec, i + 1); + snd_soc_write(codec, 0x200 + i, + pdata->gpio_init[i] & 0xffff); + } - /* Put the speakers into mono mode? */ - if (pdata->spk_mono) - reg_cache[WM8962_CLASS_D_CONTROL_2] - |= WM8962_SPK_MONO; - - /* Micbias setup, detection enable and detection - * threasholds. */ - if (pdata->mic_cfg) - snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4, - WM8962_MICDET_ENA | - WM8962_MICDET_THR_MASK | - WM8962_MICSHORT_THR_MASK | - WM8962_MICBIAS_LVL, - pdata->mic_cfg); - } + /* Put the speakers into mono mode? */ + if (pdata->spk_mono) + reg_cache[WM8962_CLASS_D_CONTROL_2] + |= WM8962_SPK_MONO; + + /* Micbias setup, detection enable and detection + * threasholds. */ + if (pdata->mic_cfg) + snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4, + WM8962_MICDET_ENA | + WM8962_MICDET_THR_MASK | + WM8962_MICSHORT_THR_MASK | + WM8962_MICBIAS_LVL, + pdata->mic_cfg); /* Latch volume update bits */ snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, @@ -3506,7 +3506,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_init_gpio(codec); if (wm8962->irq) { - if (pdata && pdata->irq_active_low) { + if (pdata->irq_active_low) { trigger = IRQF_TRIGGER_LOW; irq_pol = WM8962_IRQ_POL; } else { @@ -3603,6 +3603,10 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, init_completion(&wm8962->fll_lock); wm8962->irq = i2c->irq; + /* If platform data was supplied, update the default data in priv */ + if (pdata) + memcpy(&wm8962->pdata, pdata, sizeof(struct wm8962_pdata)); + for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; @@ -3666,7 +3670,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, goto err_enable; } - if (pdata && pdata->in4_dc_measure) { + if (wm8962->pdata.in4_dc_measure) { ret = regmap_register_patch(wm8962->regmap, wm8962_dc_measure, ARRAY_SIZE(wm8962_dc_measure)); -- cgit v1.2.1 From d74e9e7090aeb9b61e683e5abf7ca70fa18f846b Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 7 Jun 2013 11:23:27 +0800 Subject: ASoC: wm8962: Add device tree binding Document the device tree binding for the WM8962 codec, and modify the driver to extract platform data from the device tree, if present. Based on work of WM8903 by Stephen Warren Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d56dd867057d..26219ea2bbb5 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3584,6 +3584,34 @@ static const struct regmap_config wm8962_regmap = { .cache_type = REGCACHE_RBTREE, }; +static int wm8962_set_pdata_from_of(struct i2c_client *i2c, + struct wm8962_pdata *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + u32 val32; + int i; + + if (of_property_read_bool(np, "spk-mono")) + pdata->spk_mono = true; + + if (of_property_read_u32(np, "mic-cfg", &val32) >= 0) + pdata->mic_cfg = val32; + + if (of_property_read_u32_array(np, "gpio-cfg", pdata->gpio_init, + ARRAY_SIZE(pdata->gpio_init)) >= 0) + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) { + /* + * The range of GPIO register value is [0x0, 0xffff] + * While the default value of each register is 0x0 + * Any other value will be regarded as default value + */ + if (pdata->gpio_init[i] > 0xffff) + pdata->gpio_init[i] = 0x0; + } + + return 0; +} + static int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3604,8 +3632,13 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, wm8962->irq = i2c->irq; /* If platform data was supplied, update the default data in priv */ - if (pdata) + if (pdata) { memcpy(&wm8962->pdata, pdata, sizeof(struct wm8962_pdata)); + } else if (i2c->dev.of_node) { + ret = wm8962_set_pdata_from_of(i2c, &wm8962->pdata); + if (ret != 0) + return ret; + } for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; -- cgit v1.2.1 From 9e13f345887c179068bbc1f7389b7177bf88f57e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 9 Jun 2013 22:07:46 -0300 Subject: ASoC: sgtl5000: Let the codec acquire its clock On a mx6qsabrelite board the following error happens on probe: sgtl5000: probe of 0-000a failed with error -5 imx-sgtl5000 sound.13: ASoC: CODEC (null) not registered imx-sgtl5000 sound.13: snd_soc_register_card failed (-517) platform sound.13: Driver imx-sgtl5000 requests probe defer Prior to reading the codec ID we need to turn the SYS_MCLK clock, so let's enable the codec clock inside sgtl5000_i2c_probe(). Also remove the codec clock enable/disable functions from the machine driver. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/fsl/imx-sgtl5000.c | 30 +++++++----------------------- 2 files changed, 37 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8f2afb74706..2e0227bda8e0 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -114,6 +114,7 @@ struct sgtl5000_priv { struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; struct regmap *regmap; + struct clk *mclk; }; /* @@ -1522,16 +1523,28 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + sgtl5000->mclk = devm_clk_get(&client->dev, NULL); + if (IS_ERR(sgtl5000->mclk)) { + ret = PTR_ERR(sgtl5000->mclk); + dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(sgtl5000->mclk); + if (ret) + return ret; + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); if (ret) - return ret; + goto disable_clk; if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != SGTL5000_PARTID_PART_ID) { dev_err(&client->dev, "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; + ret = -ENODEV; + goto disable_clk; } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; @@ -1542,17 +1555,30 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* Ensure sgtl5000 will start with sane register values */ ret = sgtl5000_fill_defaults(sgtl5000); if (ret) - return ret; + goto disable_clk; ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto disable_clk; + + return 0; + +disable_clk: + clk_disable_unprepare(sgtl5000->mclk); return ret; } static int sgtl5000_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + struct sgtl5000_priv *sgtl5000; + sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), + GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + snd_soc_unregister_codec(&client->dev); + clk_disable_unprepare(sgtl5000->mclk); return 0; } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index a60aaa053d28..823151b7653b 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,20 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - /* assuming clock enabled by default */ - data->codec_clk = NULL; - ret = of_property_read_u32(codec_np, "clock-frequency", - &data->clk_frequency); - if (ret) { - dev_err(&codec_dev->dev, - "clock-frequency missing or invalid\n"); - goto fail; - } - } else { - data->clk_frequency = clk_get_rate(data->codec_clk); - clk_prepare_enable(data->codec_clk); - } + if (IS_ERR(data->codec_clk)) + goto fail; + + data->clk_frequency = clk_get_rate(data->codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -157,10 +147,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -170,7 +160,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = snd_soc_register_card(&data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } platform_set_drvdata(pdev, data); @@ -179,8 +169,6 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; -clk_fail: - clk_put(data->codec_clk); fail: if (ssi_np) of_node_put(ssi_np); @@ -194,10 +182,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) { struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - if (data->codec_clk) { - clk_disable_unprepare(data->codec_clk); - clk_put(data->codec_clk); - } snd_soc_unregister_card(&data->card); return 0; -- cgit v1.2.1 From 9c24b1672283644adf871244771ebf387dd73f90 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Jun 2013 16:19:58 +0100 Subject: ASoC: wm8962: Restore device state after reset in runtime resume After the device has been reset we need to repeat the same initialisation we do on probe to make sure that the device is in a known state. Tested-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 26219ea2bbb5..7a7a0567e547 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3756,6 +3756,21 @@ static int wm8962_runtime_resume(struct device *dev) wm8962_reset(wm8962); + /* SYSCLK defaults to on; make sure it is off so we can safely + * write to registers if the device is declocked. + */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA, 0); + + /* Ensure we have soft control over all registers */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + + /* Ensure that the oscillator and PLLs are disabled */ + regmap_update_bits(wm8962->regmap, WM8962_PLL2, + WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, + 0); + regcache_sync(wm8962->regmap); return 0; -- cgit v1.2.1 From 7c647af43f1517b5b2604b8a69ea72a17073e15f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 10 Jun 2013 10:24:41 -0300 Subject: ASoC: sgtl5000: Use i2c_get_clientdata() We should use i2c_get_clientdata() to get the codec private structure. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2e0227bda8e0..d441559dc92c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1571,11 +1571,7 @@ disable_clk: static int sgtl5000_i2c_remove(struct i2c_client *client) { - struct sgtl5000_priv *sgtl5000; - sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), - GFP_KERNEL); - if (!sgtl5000) - return -ENOMEM; + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); clk_disable_unprepare(sgtl5000->mclk); -- cgit v1.2.1 From b9840124d699614f1429748e43827b1fb35c1138 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 10 Jun 2013 13:26:05 -0300 Subject: ASoC: imx-sgtl5000: Use devm_clk_get() Commit 9e13f345 (ASoC: sgtl5000: Let the codec acquire its clock) removed the clk_put calls. Let's use devm_clk_get() instead, so that we do not need to call them anymore. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 823151b7653b..7a8bc1220b2e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = clk_get(&codec_dev->dev, NULL); + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) goto fail; -- cgit v1.2.1 From 8de2ae2a7f1fd71dc56d6b014029f93093e9c5d5 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 11 Jun 2013 02:43:30 +0800 Subject: ASoC: fsl: add imx-wm8962 machine driver This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 12 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 323 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 337 insertions(+) create mode 100644 sound/soc/fsl/imx-wm8962.c (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc27e5b2..aa438546c912 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8a2321..d4b4aa8b5649 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 000000000000..52a36a90f4f4 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,323 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + struct platform_device *pdev; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct device *dev = &priv->pdev->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, + "failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct device *dev = &priv->pdev->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(dev, "failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + ret = clk_prepare_enable(data->codec_clk); + if (ret) { + dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); + goto fail; + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- cgit v1.2.1 From 5e83c160d83070152a595f57a6ca7c5bb1ce16b3 Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Tue, 11 Jun 2013 09:29:07 +0530 Subject: ASoC: dwc: debug message correction. Debug message correction. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 593a3ea12d4c..2c625264223b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -396,7 +396,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " SPEAr: play supported\n"); + dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->playback.channels_max = pdata->channel; dw_i2s_dai->playback.formats = pdata->snd_fmts; @@ -404,7 +404,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "SPEAr: record supported\n"); + dev_dbg(&pdev->dev, "designware: record supported\n"); dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->capture.channels_max = pdata->channel; dw_i2s_dai->capture.formats = pdata->snd_fmts; -- cgit v1.2.1 From 22a4adf25826d1128d116dd4a313f66175c703bd Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Tue, 11 Jun 2013 09:29:08 +0530 Subject: ASoC: dwc: Folder path correction in file header. Folder path correction in file header. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 2c625264223b..70eb37a5dd16 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -1,7 +1,7 @@ /* * ALSA SoC Synopsys I2S Audio Layer * - * sound/soc/spear/designware_i2s.c + * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics * Rajeev Kumar -- cgit v1.2.1 From 1aad4e574bced05b4036e79981a7800dd275cf1c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 10 Jun 2013 22:23:53 +0800 Subject: ASoC: ssm2518: Fix trivial typo in checking tx_mask and rx_mask values Otherwise, ssm2518_set_tdm_slot() always returns error if slots != 0. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2518.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 3139a1bde295..95aed552139a 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -539,7 +539,7 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, SSM2518_SAI_CTRL1_SAI_I2S); - if (tx_mask == 0 || tx_mask != 0) + if (tx_mask == 0 || rx_mask != 0) return -EINVAL; if (slots == 1) { -- cgit v1.2.1 From f724ba3b07aa5a012b7b0be323484195b5026282 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 7 Jun 2013 13:53:04 +0200 Subject: ASoC: codecs: adau1701: factor out firmware reset Some runtime-determined constraints might need to be satisfied prior to firmware loading, so the actual download and releasing the device from reset has to be postponed. Factor it out first, so we have everything at one place. This also changes the behaviour in a way that adau1701_i2c_probe() will assert the reset line, and wait for the codec probe to release it. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 49 +++++++++++++++++++++++++++++++++------------ 1 file changed, 36 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 3fc176387351..b6b1a773bd37 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -90,6 +90,7 @@ #define ADAU1701_FIRMWARE "adau1701.bin" struct adau1701 { + int gpio_nreset; unsigned int dai_fmt; }; @@ -183,9 +184,37 @@ static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) return value; } -static int adau1701_load_firmware(struct i2c_client *client) +static void adau1701_reset(struct snd_soc_codec *codec) { - return process_sigma_firmware(client, ADAU1701_FIRMWARE); + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + + if (!gpio_is_valid(adau1701->gpio_nreset)) + return; + + gpio_set_value(adau1701->gpio_nreset, 0); + /* minimum reset time is 20ns */ + udelay(1); + gpio_set_value(adau1701->gpio_nreset, 1); + /* power-up time may be as long as 85ms */ + mdelay(85); +} + +static int adau1701_init(struct snd_soc_codec *codec) +{ + int ret; + struct i2c_client *client = to_i2c_client(codec->dev); + + adau1701_reset(codec); + + ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); + if (ret) { + dev_warn(codec->dev, "Failed to load firmware\n"); + return ret; + } + + snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + + return 0; } static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, @@ -466,15 +495,13 @@ MODULE_DEVICE_TABLE(of, adau1701_dt_ids); static int adau1701_probe(struct snd_soc_codec *codec) { int ret; - struct i2c_client *client = to_i2c_client(codec->dev); - codec->control_data = client; + codec->control_data = to_i2c_client(codec->dev); - ret = adau1701_load_firmware(client); + ret = adau1701_init(codec); if (ret) - dev_warn(codec->dev, "Failed to load firmware\n"); + return ret; - snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); return 0; @@ -524,14 +551,10 @@ static int adau1701_i2c_probe(struct i2c_client *client, "ADAU1701 Reset"); if (ret < 0) return ret; - - /* minimum reset time is 20ns */ - udelay(1); - gpio_set_value(gpio_nreset, 1); - /* power-up time may be as long as 85ms */ - mdelay(85); } + adau1701->gpio_nreset = gpio_nreset; + i2c_set_clientdata(client, adau1701); ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); -- cgit v1.2.1 From 4119c0c0c6e4508236672c3ec714da25eed783ce Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 10 Jun 2013 21:20:44 +0800 Subject: ASoC: adav80x: Select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI This driver is useless if both SPI and I2C are not configured. Thus don't build this driver if both SPI and I2C are not configured. This patch silences below build warning if both SPI and I2C are not configured. CC sound/soc/codecs/adav80x.o sound/soc/codecs/adav80x.c:842:12: warning: 'adav80x_bus_probe' defined but not used [-Wunused-function] sound/soc/codecs/adav80x.c:863:12: warning: 'adav80x_bus_remove' defined but not used [-Wunused-function] Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00e31b0..5841674b6993 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,7 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAV80X + select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- cgit v1.2.1 From e58070ee4fdf797c47cb296992ce8db3df715eca Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Mon, 10 Jun 2013 16:19:40 +0530 Subject: ASoC: Add Kconfig and Makefile to support SPEAr audio driver This patch adds Kconfig and Makefile to support SPEAr Audio driver. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/spear/Kconfig | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/spear/Makefile | 7 +++++++ 4 files changed, 48 insertions(+) create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 9e675c76436c..45eeaa9f7fec 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -51,6 +51,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 197b6ae54c8d..bc0261476d7a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -29,6 +29,7 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig new file mode 100644 index 000000000000..3b7cdadc11cc --- /dev/null +++ b/sound/soc/spear/Kconfig @@ -0,0 +1,39 @@ +config SND_SPEAR_EVM + tristate "SoC Audio support for SPEAr EVM" + select SND_DESIGNWARE_I2S + select SND_SOC_STA529 + select SND_SPEAR_SOC + help + Say Y if you want to add support for SoC audio on SPEAr + platform + +config SND_SPEAR1340_EVM + tristate "SoC Audio support for SPEAr1340 EVM" + select SND_DESIGNWARE_I2S + select SND_SOC_STA529 + select SND_SPEAR_SPDIF_OUT + select SND_SPEAR_SPDIF_IN + select SND_SOC_SPDIF + select SND_SPEAR_SOC + help + Say Y if you want to add support for SoC audio on SPEAr1340 + platform + +config SND_SPEAR_SOC + tristate "SoC Audio for the ST chip" + depends on SND_DESIGNWARE_I2S || SND_SPEAR_SPDIF_OUT || \ + SND_SPEAR_SPDIF_IN + help + Say Y or M if you want to add support for any of the audio + controllers (I2S/SPDIF). You will also need to select + the audio interface codecs to support below. + +config SND_SPEAR_SPDIF_OUT + tristate "SPEAr SPDIF Out Device Driver" + help + Say Y or M if you want to add support for SPDIF OUT driver. + +config SND_SPEAR_SPDIF_IN + tristate "SPEAr SPDIF IN Device Driver" + help + Say Y or M if you want to add support for SPDIF IN driver. diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile new file mode 100644 index 000000000000..b36512655bcf --- /dev/null +++ b/sound/soc/spear/Makefile @@ -0,0 +1,7 @@ +# SPEAR Platform Support +obj-$(CONFIG_SND_SPEAR_SOC) += spear_pcm.o +obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += spdif_in.o +obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += spdif_out.o + +# SPEAR Machine Support +obj-$(CONFIG_SND_SPEAR_EVM) += spear_evb.o -- cgit v1.2.1 From f3fe53dd975306903be3616c87865a87a52fb20e Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Wed, 12 Jun 2013 09:57:57 +0200 Subject: ASoC: ux500: Move DMA parameters into ux500_msp Move struct ux500_msp_dma_params declaration from ux500_msp_i2s_drvdata to ux500_msp, this saves some confusing pointer passing and allows to access all DMA configuration fields from ux500_msp_i2s. Signed-off-by: Fabio Baltieri Acked-by: Linus Walleij Acked-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 11 ++++------- sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 10 ++++++---- sound/soc/ux500/ux500_msp_i2s.h | 14 +++++++------- 4 files changed, 17 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 7d5fc1328523..c6fb5cce980e 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -658,14 +658,11 @@ static int ux500_msp_dai_probe(struct snd_soc_dai *dai) { struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); - drvdata->playback_dma_data.dma_cfg = drvdata->msp->dma_cfg_tx; - drvdata->capture_dma_data.dma_cfg = drvdata->msp->dma_cfg_rx; + dai->playback_dma_data = &drvdata->msp->playback_dma_data; + dai->capture_dma_data = &drvdata->msp->capture_dma_data; - dai->playback_dma_data = &drvdata->playback_dma_data; - dai->capture_dma_data = &drvdata->capture_dma_data; - - drvdata->playback_dma_data.data_size = drvdata->slot_width; - drvdata->capture_dma_data.data_size = drvdata->slot_width; + drvdata->msp->playback_dma_data.data_size = drvdata->slot_width; + drvdata->msp->capture_dma_data.data_size = drvdata->slot_width; return 0; } diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index c7212825fe4c..312ae535e351 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -51,8 +51,6 @@ enum ux500_msp_clock_id { struct ux500_msp_i2s_drvdata { struct ux500_msp *msp; struct regulator *reg_vape; - struct ux500_msp_dma_params playback_dma_data; - struct ux500_msp_dma_params capture_dma_data; unsigned int fmt; unsigned int tx_mask; unsigned int rx_mask; diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index cba0e86833e9..14a4a5bb60fc 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -367,12 +367,14 @@ static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) } /* Make sure the correct DMA-directions are configured */ - if ((config->direction & MSP_DIR_RX) && (!msp->dma_cfg_rx)) { + if ((config->direction & MSP_DIR_RX) && + !msp->capture_dma_data.dma_cfg) { dev_err(msp->dev, "%s: ERROR: MSP RX-mode is not configured!", __func__); return -EINVAL; } - if ((config->direction == MSP_DIR_TX) && (!msp->dma_cfg_tx)) { + if ((config->direction == MSP_DIR_TX) && + !msp->playback_dma_data.dma_cfg) { dev_err(msp->dev, "%s: ERROR: MSP TX-mode is not configured!", __func__); return -EINVAL; @@ -673,8 +675,8 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->id = platform_data->id; msp->dev = &pdev->dev; - msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; - msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; + msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx; + msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) { diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 189a3751993b..879617147fc8 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -468,12 +468,17 @@ struct ux500_msp_config { unsigned int iodelay; }; +struct ux500_msp_dma_params { + unsigned int data_size; + struct stedma40_chan_cfg *dma_cfg; +}; + struct ux500_msp { enum msp_i2s_id id; void __iomem *registers; struct device *dev; - struct stedma40_chan_cfg *dma_cfg_rx; - struct stedma40_chan_cfg *dma_cfg_tx; + struct ux500_msp_dma_params playback_dma_data; + struct ux500_msp_dma_params capture_dma_data; enum msp_state msp_state; int def_elem_len; unsigned int dir_busy; @@ -481,11 +486,6 @@ struct ux500_msp { unsigned int f_bitclk; }; -struct ux500_msp_dma_params { - unsigned int data_size; - struct stedma40_chan_cfg *dma_cfg; -}; - struct msp_i2s_platform_data; int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, -- cgit v1.2.1 From 20413113ffdd8c56b2a985ca8195d9c91e9c602b Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Wed, 12 Jun 2013 09:57:58 +0200 Subject: ASoC: ux500: Set DMA address during device init Add a field with the tx/rx register address to the DMA parameters structure, and set it to the correct address during device initialization. This address used to be hardcoded in the DMA controller driver, it now needs to be explicitly figured out by the device driver. Signed-off-by: Fabio Baltieri Acked-by: Linus Walleij Acked-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.c | 3 +++ sound/soc/ux500/ux500_msp_i2s.h | 1 + 2 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 14a4a5bb60fc..1ca8b08ae993 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -685,6 +685,9 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, return -ENOMEM; } + msp->playback_dma_data.tx_rx_addr = res->start + MSP_DR; + msp->capture_dma_data.tx_rx_addr = res->start + MSP_DR; + msp->registers = devm_ioremap(&pdev->dev, res->start, resource_size(res)); if (msp->registers == NULL) { diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 879617147fc8..258d0bcee0bd 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -470,6 +470,7 @@ struct ux500_msp_config { struct ux500_msp_dma_params { unsigned int data_size; + dma_addr_t tx_rx_addr; struct stedma40_chan_cfg *dma_cfg; }; -- cgit v1.2.1 From eef6473ff3ce2383febebd2e799beceaece9adda Mon Sep 17 00:00:00 2001 From: Fabio Baltieri Date: Wed, 12 Jun 2013 09:57:59 +0200 Subject: ASoC: ux500: Add DMA slave config prepare routine Implement a DMA slave config prepare routine, as until now the MSP driver depended on the DMA controller completing the channel configuration on its own, but this is not the case anymore since the recent DMA driver updates. Signed-off-by: Fabio Baltieri Acked-by: Linus Walleij Acked-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_pcm.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index b6e5ae277299..5f01c19776bf 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -103,10 +103,40 @@ static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, return snd_dmaengine_pcm_request_channel(stedma40_filter, dma_cfg); } +static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ux500_msp_dma_params *dma_params; + struct stedma40_chan_cfg *dma_cfg; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_cfg = dma_params->dma_cfg; + + ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); + if (ret) + return ret; + + slave_config->dst_maxburst = 4; + slave_config->dst_addr_width = dma_cfg->dst_info.data_width; + slave_config->src_maxburst = 4; + slave_config->src_addr_width = dma_cfg->src_info.data_width; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + slave_config->dst_addr = dma_params->tx_rx_addr; + else + slave_config->src_addr = dma_params->tx_rx_addr; + + return 0; +} + static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .pcm_hardware = &ux500_pcm_hw, .compat_request_channel = ux500_pcm_request_chan, .prealloc_buffer_size = 128 * 1024, + .prepare_slave_config = ux500_pcm_prepare_slave_config, }; int ux500_pcm_register_platform(struct platform_device *pdev) -- cgit v1.2.1 From 997b05203b0a710e11f9b2732bef2d2fdc1d824b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 11 Jun 2013 13:10:16 +0800 Subject: ASoC: add RT5640 CODEC driver This patch adds the ALC5640 codec driver. Signed-off-by: Stephen Warren Signed-off-by: Bard Liao Tested-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5640.c | 2092 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2092 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 4190 insertions(+) create mode 100644 sound/soc/codecs/rt5640.c create mode 100644 sound/soc/codecs/rt5640.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00e31b0..04c87f77b7e8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C + select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -296,6 +297,9 @@ config SND_SOC_PCM3008 config SND_SOC_RT5631 tristate +config SND_SOC_RT5640 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9a1f4c..732c4a7bd6ac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -44,6 +44,7 @@ snd-soc-ml26124-objs := ml26124.o snd-soc-omap-hdmi-codec-objs := omap-hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o +snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o +obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c new file mode 100644 index 000000000000..288c17cd6023 --- /dev/null +++ b/sound/soc/codecs/rt5640.c @@ -0,0 +1,2092 @@ +/* + * rt5640.c -- RT5640 ALSA SoC audio codec driver + * + * Copyright 2011 Realtek Semiconductor Corp. + * Author: Johnny Hsu + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rt5640.h" + +#define RT5640_DEVICE_ID 0x6231 + +#define RT5640_PR_RANGE_BASE (0xff + 1) +#define RT5640_PR_SPACING 0x100 + +#define RT5640_PR_BASE (RT5640_PR_RANGE_BASE + (0 * RT5640_PR_SPACING)) + +static const struct regmap_range_cfg rt5640_ranges[] = { + { .name = "PR", .range_min = RT5640_PR_BASE, + .range_max = RT5640_PR_BASE + 0xb4, + .selector_reg = RT5640_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5640_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + {RT5640_PR_BASE + 0x3d, 0x3600}, + {RT5640_PR_BASE + 0x1c, 0x0D21}, + {RT5640_PR_BASE + 0x1b, 0x0000}, + {RT5640_PR_BASE + 0x12, 0x0aa8}, + {RT5640_PR_BASE + 0x14, 0x0aaa}, + {RT5640_PR_BASE + 0x20, 0x6110}, + {RT5640_PR_BASE + 0x21, 0xe0e0}, + {RT5640_PR_BASE + 0x23, 0x1804}, +}; +#define RT5640_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5640_reg[RT5640_VENDOR_ID2 + 1] = { + { 0x00, 0x000e }, + { 0x01, 0xc8c8 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x04, 0x8000 }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0000 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x7060 }, + { 0x28, 0x7070 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5454 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaa00 }, + { 0x2d, 0x0000 }, + { 0x2e, 0xa000 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x47, 0x003e }, + { 0x48, 0xf800 }, + { 0x49, 0x3800 }, + { 0x4a, 0x0004 }, + { 0x4c, 0xfc00 }, + { 0x4d, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x53, 0xf000 }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6c, 0x0000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1114 }, + { 0x74, 0x0c00 }, + { 0x75, 0x1d00 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0008 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0600 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c00 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb8, 0x034b }, + { 0xb9, 0x0066 }, + { 0xba, 0x000b }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0400 }, + { 0xc2, 0x0000 }, + { 0xc4, 0x0000 }, + { 0xc5, 0x0000 }, + { 0xc6, 0x2000 }, + { 0xc8, 0x0000 }, + { 0xc9, 0x0000 }, + { 0xca, 0x0000 }, + { 0xcb, 0x0000 }, + { 0xcc, 0x0000 }, + { 0xcf, 0x0013 }, + { 0xd0, 0x0680 }, + { 0xd1, 0x1c17 }, + { 0xd2, 0x8c00 }, + { 0xd3, 0xaa20 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6231 }, +}; + +static int rt5640_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5640_RESET, 0); +} + +static bool rt5640_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5640_ranges); i++) + if ((reg >= rt5640_ranges[i].window_start && + reg <= rt5640_ranges[i].window_start + + rt5640_ranges[i].window_len) || + (reg >= rt5640_ranges[i].range_min && + reg <= rt5640_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5640_RESET: + case RT5640_ASRC_5: + case RT5640_EQ_CTRL1: + case RT5640_DRC_AGC_1: + case RT5640_ANC_CTRL1: + case RT5640_IRQ_CTRL2: + case RT5640_INT_IRQ_ST: + case RT5640_DSP_CTRL2: + case RT5640_DSP_CTRL3: + case RT5640_PRIV_INDEX: + case RT5640_PRIV_DATA: + case RT5640_PGM_REG_ARR1: + case RT5640_PGM_REG_ARR3: + case RT5640_VENDOR_ID: + case RT5640_VENDOR_ID1: + case RT5640_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5640_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5640_ranges); i++) + if ((reg >= rt5640_ranges[i].window_start && + reg <= rt5640_ranges[i].window_start + + rt5640_ranges[i].window_len) || + (reg >= rt5640_ranges[i].range_min && + reg <= rt5640_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5640_RESET: + case RT5640_SPK_VOL: + case RT5640_HP_VOL: + case RT5640_OUTPUT: + case RT5640_MONO_OUT: + case RT5640_IN1_IN2: + case RT5640_IN3_IN4: + case RT5640_INL_INR_VOL: + case RT5640_DAC1_DIG_VOL: + case RT5640_DAC2_DIG_VOL: + case RT5640_DAC2_CTRL: + case RT5640_ADC_DIG_VOL: + case RT5640_ADC_DATA: + case RT5640_ADC_BST_VOL: + case RT5640_STO_ADC_MIXER: + case RT5640_MONO_ADC_MIXER: + case RT5640_AD_DA_MIXER: + case RT5640_STO_DAC_MIXER: + case RT5640_MONO_DAC_MIXER: + case RT5640_DIG_MIXER: + case RT5640_DSP_PATH1: + case RT5640_DSP_PATH2: + case RT5640_DIG_INF_DATA: + case RT5640_REC_L1_MIXER: + case RT5640_REC_L2_MIXER: + case RT5640_REC_R1_MIXER: + case RT5640_REC_R2_MIXER: + case RT5640_HPO_MIXER: + case RT5640_SPK_L_MIXER: + case RT5640_SPK_R_MIXER: + case RT5640_SPO_L_MIXER: + case RT5640_SPO_R_MIXER: + case RT5640_SPO_CLSD_RATIO: + case RT5640_MONO_MIXER: + case RT5640_OUT_L1_MIXER: + case RT5640_OUT_L2_MIXER: + case RT5640_OUT_L3_MIXER: + case RT5640_OUT_R1_MIXER: + case RT5640_OUT_R2_MIXER: + case RT5640_OUT_R3_MIXER: + case RT5640_LOUT_MIXER: + case RT5640_PWR_DIG1: + case RT5640_PWR_DIG2: + case RT5640_PWR_ANLG1: + case RT5640_PWR_ANLG2: + case RT5640_PWR_MIXER: + case RT5640_PWR_VOL: + case RT5640_PRIV_INDEX: + case RT5640_PRIV_DATA: + case RT5640_I2S1_SDP: + case RT5640_I2S2_SDP: + case RT5640_ADDA_CLK1: + case RT5640_ADDA_CLK2: + case RT5640_DMIC: + case RT5640_GLB_CLK: + case RT5640_PLL_CTRL1: + case RT5640_PLL_CTRL2: + case RT5640_ASRC_1: + case RT5640_ASRC_2: + case RT5640_ASRC_3: + case RT5640_ASRC_4: + case RT5640_ASRC_5: + case RT5640_HP_OVCD: + case RT5640_CLS_D_OVCD: + case RT5640_CLS_D_OUT: + case RT5640_DEPOP_M1: + case RT5640_DEPOP_M2: + case RT5640_DEPOP_M3: + case RT5640_CHARGE_PUMP: + case RT5640_PV_DET_SPK_G: + case RT5640_MICBIAS: + case RT5640_EQ_CTRL1: + case RT5640_EQ_CTRL2: + case RT5640_WIND_FILTER: + case RT5640_DRC_AGC_1: + case RT5640_DRC_AGC_2: + case RT5640_DRC_AGC_3: + case RT5640_SVOL_ZC: + case RT5640_ANC_CTRL1: + case RT5640_ANC_CTRL2: + case RT5640_ANC_CTRL3: + case RT5640_JD_CTRL: + case RT5640_ANC_JD: + case RT5640_IRQ_CTRL1: + case RT5640_IRQ_CTRL2: + case RT5640_INT_IRQ_ST: + case RT5640_GPIO_CTRL1: + case RT5640_GPIO_CTRL2: + case RT5640_GPIO_CTRL3: + case RT5640_DSP_CTRL1: + case RT5640_DSP_CTRL2: + case RT5640_DSP_CTRL3: + case RT5640_DSP_CTRL4: + case RT5640_PGM_REG_ARR1: + case RT5640_PGM_REG_ARR2: + case RT5640_PGM_REG_ARR3: + case RT5640_PGM_REG_ARR4: + case RT5640_PGM_REG_ARR5: + case RT5640_SCB_FUNC: + case RT5640_SCB_CTRL: + case RT5640_BASE_BACK: + case RT5640_MP3_PLUS1: + case RT5640_MP3_PLUS2: + case RT5640_3D_HP: + case RT5640_ADJ_HPF: + case RT5640_HP_CALIB_AMP_DET: + case RT5640_HP_CALIB2: + case RT5640_SV_ZCD1: + case RT5640_SV_ZCD2: + case RT5640_DUMMY1: + case RT5640_DUMMY2: + case RT5640_DUMMY3: + case RT5640_VENDOR_ID: + case RT5640_VENDOR_ID1: + case RT5640_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5640_data_select[] = { + "Normal", "left copy to right", "right copy to left", "Swap"}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); + +/* Class D speaker gain ratio */ +static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", + "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, + RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); + +static const struct snd_kcontrol_new rt5640_snd_controls[] = { + /* Speaker Output Volume */ + SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, + RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, + RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5640_OUTPUT, + RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5640_OUTPUT, + RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, + RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, + RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, + RT5640_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, + RT5640_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, + RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5640_ADC_DIG_VOL, + RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5640_ADC_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 127, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5640_ADC_DATA, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 127, 0, adc_vol_tlv), + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("ADC Boost Gain", RT5640_ADC_BST_VOL, + RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + /* Class D speaker gain ratio */ + SOC_ENUM("Class D SPK Ratio Control", rt5640_clsd_spk_ratio_enum), + + SOC_ENUM("ADC IF1 Data Switch", rt5640_if1_adc_enum), + SOC_ENUM("DAC IF1 Data Switch", rt5640_if1_dac_enum), + SOC_ENUM("ADC IF2 Data Switch", rt5640_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int idx = -EINVAL, i; + int rate, red, bound, temp; + + rate = rt5640->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5640_DMIC, RT5640_DMIC_CLK_MASK, + idx << RT5640_DMIC_CLK_SFT); + return idx; +} + +static int check_sysclk1_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5640_GLB_CLK); + val &= RT5640_SCLK_SRC_MASK; + if (val == RT5640_SCLK_SRC_PLL1 || val == RT5640_SCLK_SRC_PLL1T) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5640_sto_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_sto_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_STO_ADC_MIXER, + RT5640_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5640_MONO_ADC_MIXER, + RT5640_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5640_AD_DA_MIXER, + RT5640_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5640_AD_DA_MIXER, + RT5640_M_IF1_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5640_AD_DA_MIXER, + RT5640_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5640_AD_DA_MIXER, + RT5640_M_IF1_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("ANC Switch", RT5640_STO_DAC_MIXER, + RT5640_M_ANC_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("ANC Switch", RT5640_STO_DAC_MIXER, + RT5640_M_ANC_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_MONO_DAC_MIXER, + RT5640_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dig_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_DIG_MIXER, + RT5640_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_DIG_MIXER, + RT5640_M_DAC_L2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_dig_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_DIG_MIXER, + RT5640_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_DIG_MIXER, + RT5640_M_DAC_R2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5640_rec_l_mix[] = { + SOC_DAPM_SINGLE("HPOL Switch", RT5640_REC_L2_MIXER, + RT5640_M_HP_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER, + RT5640_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST4_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5640_REC_L2_MIXER, + RT5640_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_rec_r_mix[] = { + SOC_DAPM_SINGLE("HPOR Switch", RT5640_REC_R2_MIXER, + RT5640_M_HP_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER, + RT5640_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST4_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5640_REC_R2_MIXER, + RT5640_M_OM_R_RM_R_SFT, 1, 1), +}; + +/* Analog Output Mixer */ +static const struct snd_kcontrol_new rt5640_spk_l_mix[] = { + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_SPK_L_MIXER, + RT5640_M_RM_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_SPK_L_MIXER, + RT5640_M_IN_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_SPK_L_MIXER, + RT5640_M_DAC_L1_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_SPK_L_MIXER, + RT5640_M_DAC_L2_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5640_SPK_L_MIXER, + RT5640_M_OM_L_SM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_spk_r_mix[] = { + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_SPK_R_MIXER, + RT5640_M_RM_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_SPK_R_MIXER, + RT5640_M_IN_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPK_R_MIXER, + RT5640_M_DAC_R1_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_SPK_R_MIXER, + RT5640_M_DAC_R2_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5640_SPK_R_MIXER, + RT5640_M_OM_R_SM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_out_l_mix[] = { + SOC_DAPM_SINGLE("SPK MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_SM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_R2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_out_r_mix[] = { + SOC_DAPM_SINGLE("SPK MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_SM_L_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_L2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, + RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_SPO_L_MIXER, + RT5640_M_DAC_L1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5640_SPO_L_MIXER, + RT5640_M_SV_R_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL L Switch", RT5640_SPO_L_MIXER, + RT5640_M_SV_L_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_SPO_L_MIXER, + RT5640_M_BST1_SPM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_spo_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_R_MIXER, + RT5640_M_DAC_R1_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5640_SPO_R_MIXER, + RT5640_M_SV_R_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_SPO_R_MIXER, + RT5640_M_BST1_SPM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC2 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC2_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, + RT5640_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_LOUT_MIXER, + RT5640_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5640_LOUT_MIXER, + RT5640_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5640_LOUT_MIXER, + RT5640_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5640_mono_mix[] = { + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_MONO_MIXER, + RT5640_M_DAC_R2_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_MONO_MIXER, + RT5640_M_DAC_L2_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5640_MONO_MIXER, + RT5640_M_OV_R_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5640_MONO_MIXER, + RT5640_M_OV_L_MM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_MONO_MIXER, + RT5640_M_BST1_MM_SFT, 1, 1), +}; + +/* INL/R source */ +static const char * const rt5640_inl_src[] = { + "IN2P", "MONOP" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_inl_enum, RT5640_INL_INR_VOL, + RT5640_INL_SEL_SFT, rt5640_inl_src); + +static const struct snd_kcontrol_new rt5640_inl_mux = + SOC_DAPM_ENUM("INL source", rt5640_inl_enum); + +static const char * const rt5640_inr_src[] = { + "IN2N", "MONON" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_inr_enum, RT5640_INL_INR_VOL, + RT5640_INR_SEL_SFT, rt5640_inr_src); + +static const struct snd_kcontrol_new rt5640_inr_mux = + SOC_DAPM_ENUM("INR source", rt5640_inr_enum); + +/* Stereo ADC source */ +static const char * const rt5640_stereo_adc1_src[] = { + "DIG MIX", "ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5640_sto_adc_1_mux = + SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5640_stereo_adc1_enum); + +static const char * const rt5640_stereo_adc2_src[] = { + "DMIC1", "DMIC2", "DIG MIX" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5640_sto_adc_2_mux = + SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5640_stereo_adc2_enum); + +/* Mono ADC source */ +static const char * const rt5640_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADCL" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5640_mono_adc_l1_enum); + +static const char * const rt5640_mono_adc_l2_src[] = { + "DMIC L1", "DMIC L2", "Mono DAC MIXL" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5640_mono_adc_l2_enum); + +static const char * const rt5640_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADCR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5640_mono_adc_r1_enum); + +static const char * const rt5640_mono_adc_r2_src[] = { + "DMIC R1", "DMIC R2", "Mono DAC MIXR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5640_mono_adc_r2_enum); + +/* DAC2 channel source */ +static const char * const rt5640_dac_l2_src[] = { + "IF2", "Base L/R" +}; + +static int rt5640_dac_l2_values[] = { + 0, + 3, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt5640_dac_l2_enum, RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT, + 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values); + +static const struct snd_kcontrol_new rt5640_dac_l2_mux = + SOC_DAPM_VALUE_ENUM("DAC2 left channel source", rt5640_dac_l2_enum); + +static const char * const rt5640_dac_r2_src[] = { + "IF2", +}; + +static int rt5640_dac_r2_values[] = { + 0, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt5640_dac_r2_enum, RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT, + 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values); + +static const struct snd_kcontrol_new rt5640_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 right channel source", rt5640_dac_r2_enum); + +/* digital interface and iis interface map */ +static const char * const rt5640_dai_iis_map[] = { + "1:1|2:2", "1:2|2:1", "1:1|2:1", "1:2|2:2" +}; + +static int rt5640_dai_iis_map_values[] = { + 0, + 5, + 6, + 7, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt5640_dai_iis_map_enum, RT5640_I2S1_SDP, RT5640_I2S_IF_SFT, + 0x7, rt5640_dai_iis_map, rt5640_dai_iis_map_values); + +static const struct snd_kcontrol_new rt5640_dai_mux = + SOC_DAPM_VALUE_ENUM("DAI select", rt5640_dai_iis_map_enum); + +/* SDI select */ +static const char * const rt5640_sdi_sel[] = { + "IF1", "IF2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5640_sdi_sel_enum, RT5640_I2S2_SDP, + RT5640_I2S2_SDI_SFT, rt5640_sdi_sel); + +static const struct snd_kcontrol_new rt5640_sdi_mux = + SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); + +static int spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, + 0x0001, 0x0001); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, + 0xf000, 0xf000); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, + 0xf000, 0x0000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, + 0x0001, 0x0000); + break; + + default: + return 0; + } + return 0; +} + +static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK | RT5640_GP3_PIN_MASK, + RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP3_PIN_DMIC1_SDA); + snd_soc_update_bits(codec, RT5640_DMIC, + RT5640_DMIC_1L_LH_MASK | RT5640_DMIC_1R_LH_MASK | + RT5640_DMIC_1_DP_MASK, + RT5640_DMIC_1L_LH_FALLING | RT5640_DMIC_1R_LH_RISING | + RT5640_DMIC_1_DP_IN1P); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK | RT5640_GP4_PIN_MASK, + RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP4_PIN_DMIC2_SDA); + snd_soc_update_bits(codec, RT5640_DMIC, + RT5640_DMIC_2L_LH_MASK | RT5640_DMIC_2R_LH_MASK | + RT5640_DMIC_2_DP_MASK, + RT5640_DMIC_2L_LH_FALLING | RT5640_DMIC_2R_LH_RISING | + RT5640_DMIC_2_DP_IN1N); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, + RT5640_PWR_PLL_BIT, 0, NULL, 0), + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("LDO2", RT5640_PWR_ANLG1, + RT5640_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5640_PWR_ANLG2, + RT5640_PWR_MB1_BIT, 0, 0, 0), + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5640_DMIC, + RT5640_DMIC_1_EN_SFT, 0, rt5640_set_dmic1_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5640_DMIC, + RT5640_DMIC_2_EN_SFT, 0, rt5640_set_dmic2_event, + SND_SOC_DAPM_PRE_PMU), + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5640_PWR_ANLG2, + RT5640_PWR_BST1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2, + RT5640_PWR_BST4_BIT, 0, NULL, 0), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL, + RT5640_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL, + RT5640_PWR_IN_R_BIT, 0, NULL, 0), + /* IN Mux */ + SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux), + SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux), + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0, + rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5640_PWR_MIXER, RT5640_PWR_RM_R_BIT, 0, + rt5640_rec_r_mix, ARRAY_SIZE(rt5640_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, RT5640_PWR_DIG1, + RT5640_PWR_ADC_L_BIT, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, RT5640_PWR_DIG1, + RT5640_PWR_ADC_R_BIT, 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("Stereo Filter", RT5640_PWR_DIG2, + RT5640_PWR_ADC_SF_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_adc_l_mix, ARRAY_SIZE(rt5640_sto_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_adc_r_mix, ARRAY_SIZE(rt5640_sto_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("Mono Left Filter", RT5640_PWR_DIG2, + RT5640_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_mono_adc_l_mix, ARRAY_SIZE(rt5640_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("Mono Right Filter", RT5640_PWR_DIG2, + RT5640_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_mono_adc_r_mix, ARRAY_SIZE(rt5640_mono_adc_r_mix)), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5640_PWR_DIG1, + RT5640_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5640_PWR_DIG1, + RT5640_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("DAI1 RX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("SDI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_sdi_mux), + SND_SOC_DAPM_MUX("DAI2 RX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dai_mux), + SND_SOC_DAPM_MUX("SDI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5640_sdi_mux), + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5640_dac_r2_mux), + /* DAC Mixer */ + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_mono_dac_r_mix, ARRAY_SIZE(rt5640_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DIG MIXL", SND_SOC_NOPM, 0, 0, + rt5640_dig_l_mix, ARRAY_SIZE(rt5640_dig_l_mix)), + SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, + rt5640_dig_r_mix, ARRAY_SIZE(rt5640_dig_r_mix)), + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0), + /* SPK/OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, + 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), + SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, + 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, + RT5640_PWR_SV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKVOL R", RT5640_PWR_VOL, + RT5640_PWR_SV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL L", RT5640_PWR_VOL, + RT5640_PWR_OV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL R", RT5640_PWR_VOL, + RT5640_PWR_OV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL L", RT5640_PWR_VOL, + RT5640_PWR_HV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL R", RT5640_PWR_VOL, + RT5640_PWR_HV_R_BIT, 0, NULL, 0), + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPOL MIX", SND_SOC_NOPM, 0, + 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), + SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, + 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, + rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, + SND_SOC_NOPM, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + RT5640_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + RT5640_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, + SND_SOC_NOPM, 0, spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("MONOP"), + SND_SOC_DAPM_OUTPUT("MONON"), +}; + +static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { + {"IN1P", NULL, "LDO2"}, + {"IN2P", NULL, "LDO2"}, + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + {"DMIC L2", NULL, "DMIC2"}, + {"DMIC R2", NULL, "DMIC2"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + + {"INL VOL", NULL, "IN2P"}, + {"INR VOL", NULL, "IN2N"}, + + {"RECMIXL", "HPOL Switch", "HPOL"}, + {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST2 Switch", "BST2"}, + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, + + {"RECMIXR", "HPOR Switch", "HPOR"}, + {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + {"RECMIXR", "BST1 Switch", "BST1"}, + {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC L2", NULL, "DMIC CLK"}, + {"DMIC L2", NULL, "DMIC2 Power"}, + {"DMIC R2", NULL, "DMIC CLK"}, + {"DMIC R2", NULL, "DMIC2 Power"}, + + {"Stereo ADC L2 Mux", "DMIC1", "DMIC L1"}, + {"Stereo ADC L2 Mux", "DMIC2", "DMIC L2"}, + {"Stereo ADC L2 Mux", "DIG MIX", "DIG MIXL"}, + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DIG MIX", "DIG MIXL"}, + + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DIG MIX", "DIG MIXR"}, + {"Stereo ADC R2 Mux", "DMIC1", "DMIC R1"}, + {"Stereo ADC R2 Mux", "DMIC2", "DMIC R2"}, + {"Stereo ADC R2 Mux", "DIG MIX", "DIG MIXR"}, + + {"Mono ADC L2 Mux", "DMIC L1", "DMIC L1"}, + {"Mono ADC L2 Mux", "DMIC L2", "DMIC L2"}, + {"Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "ADCL", "ADC L"}, + + {"Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + {"Mono ADC R1 Mux", "ADCR", "ADC R"}, + {"Mono ADC R2 Mux", "DMIC R1", "DMIC R1"}, + {"Mono ADC R2 Mux", "DMIC R2", "DMIC R2"}, + {"Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + + {"Stereo ADC MIXL", "ADC1 Switch", "Stereo ADC L1 Mux"}, + {"Stereo ADC MIXL", "ADC2 Switch", "Stereo ADC L2 Mux"}, + {"Stereo ADC MIXL", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", check_sysclk1_source}, + + {"Stereo ADC MIXR", "ADC1 Switch", "Stereo ADC R1 Mux"}, + {"Stereo ADC MIXR", "ADC2 Switch", "Stereo ADC R2 Mux"}, + {"Stereo ADC MIXR", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", check_sysclk1_source}, + + {"Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux"}, + {"Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux"}, + {"Mono ADC MIXL", NULL, "Mono Left Filter"}, + {"Mono Left Filter", NULL, "PLL1", check_sysclk1_source}, + + {"Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux"}, + {"Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux"}, + {"Mono ADC MIXR", NULL, "Mono Right Filter"}, + {"Mono Right Filter", NULL, "PLL1", check_sysclk1_source}, + + {"IF2 ADC L", NULL, "Mono ADC MIXL"}, + {"IF2 ADC R", NULL, "Mono ADC MIXR"}, + {"IF1 ADC L", NULL, "Stereo ADC MIXL"}, + {"IF1 ADC R", NULL, "Stereo ADC MIXR"}, + + {"IF1 ADC", NULL, "I2S1"}, + {"IF1 ADC", NULL, "IF1 ADC L"}, + {"IF1 ADC", NULL, "IF1 ADC R"}, + {"IF2 ADC", NULL, "I2S2"}, + {"IF2 ADC", NULL, "IF2 ADC L"}, + {"IF2 ADC", NULL, "IF2 ADC R"}, + + {"DAI1 TX Mux", "1:1|2:2", "IF1 ADC"}, + {"DAI1 TX Mux", "1:2|2:1", "IF2 ADC"}, + {"DAI1 IF1 Mux", "1:1|2:1", "IF1 ADC"}, + {"DAI1 IF2 Mux", "1:1|2:1", "IF2 ADC"}, + {"SDI1 TX Mux", "IF1", "DAI1 IF1 Mux"}, + {"SDI1 TX Mux", "IF2", "DAI1 IF2 Mux"}, + + {"DAI2 TX Mux", "1:2|2:1", "IF1 ADC"}, + {"DAI2 TX Mux", "1:1|2:2", "IF2 ADC"}, + {"DAI2 IF1 Mux", "1:2|2:2", "IF1 ADC"}, + {"DAI2 IF2 Mux", "1:2|2:2", "IF2 ADC"}, + {"SDI2 TX Mux", "IF1", "DAI2 IF1 Mux"}, + {"SDI2 TX Mux", "IF2", "DAI2 IF2 Mux"}, + + {"AIF1TX", NULL, "DAI1 TX Mux"}, + {"AIF1TX", NULL, "SDI1 TX Mux"}, + {"AIF2TX", NULL, "DAI2 TX Mux"}, + {"AIF2TX", NULL, "SDI2 TX Mux"}, + + {"DAI1 RX Mux", "1:1|2:2", "AIF1RX"}, + {"DAI1 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI1 RX Mux", "1:2|2:1", "AIF2RX"}, + {"DAI1 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"DAI2 RX Mux", "1:2|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:2", "AIF2RX"}, + {"DAI2 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"IF1 DAC", NULL, "I2S1"}, + {"IF1 DAC", NULL, "DAI1 RX Mux"}, + {"IF2 DAC", NULL, "I2S2"}, + {"IF2 DAC", NULL, "DAI2 RX Mux"}, + + {"IF1 DAC L", NULL, "IF1 DAC"}, + {"IF1 DAC R", NULL, "IF1 DAC"}, + {"IF2 DAC L", NULL, "IF2 DAC"}, + {"IF2 DAC R", NULL, "IF2 DAC"}, + + {"DAC MIXL", "Stereo ADC Switch", "Stereo ADC MIXL"}, + {"DAC MIXL", "INF1 Switch", "IF1 DAC L"}, + {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, + {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, + + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + + {"DAC L1", NULL, "Stereo DAC MIXL"}, + {"DAC L1", NULL, "PLL1", check_sysclk1_source}, + {"DAC R1", NULL, "Stereo DAC MIXR"}, + {"DAC R1", NULL, "PLL1", check_sysclk1_source}, + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", check_sysclk1_source}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", check_sysclk1_source}, + + {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, + {"SPK MIXL", "INL Switch", "INL VOL"}, + {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, + {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, + {"SPK MIXR", "INR Switch", "INR VOL"}, + {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXL", "BST1 Switch", "BST1"}, + {"OUT MIXL", "INL Switch", "INL VOL"}, + {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, + + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + {"OUT MIXR", "BST2 Switch", "BST2"}, + {"OUT MIXR", "BST1 Switch", "BST1"}, + {"OUT MIXR", "INR Switch", "INR VOL"}, + {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, + + {"SPKVOL L", NULL, "SPK MIXL"}, + {"SPKVOL R", NULL, "SPK MIXR"}, + {"HPOVOL L", NULL, "OUT MIXL"}, + {"HPOVOL R", NULL, "OUT MIXR"}, + {"OUTVOL L", NULL, "OUT MIXL"}, + {"OUTVOL R", NULL, "OUT MIXR"}, + + {"SPOL MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOL MIX", "DAC L1 Switch", "DAC L1"}, + {"SPOL MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOL MIX", "SPKVOL L Switch", "SPKVOL L"}, + {"SPOL MIX", "BST1 Switch", "BST1"}, + {"SPOR MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOR MIX", "BST1 Switch", "BST1"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, + {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, + {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + + {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, + {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, + {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + + {"HP L Amp", NULL, "HPO MIX L"}, + {"HP R Amp", NULL, "HPO MIX R"}, + + {"SPOLP", NULL, "SPOL MIX"}, + {"SPOLN", NULL, "SPOL MIX"}, + {"SPORP", NULL, "SPOR MIX"}, + {"SPORN", NULL, "SPOR MIX"}, + + {"SPOLP", NULL, "Improve SPK Amp Drv"}, + {"SPOLN", NULL, "Improve SPK Amp Drv"}, + {"SPORP", NULL, "Improve SPK Amp Drv"}, + {"SPORN", NULL, "Improve SPK Amp Drv"}, + + {"HPOL", NULL, "Improve HP Amp Drv"}, + {"HPOR", NULL, "Improve HP Amp Drv"}, + + {"HPOL", NULL, "HP L Amp"}, + {"HPOR", NULL, "HP R Amp"}, + {"LOUTL", NULL, "LOUT MIX"}, + {"LOUTR", NULL, "LOUT MIX"}, + {"MONOP", NULL, "Mono MIX"}, + {"MONON", NULL, "Mono MIX"}, + {"MONOP", NULL, "Improve MONO Amp Drv"}, +}; + +static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) +{ + int ret = 0, val; + + if (codec == NULL) + return -EINVAL; + + val = snd_soc_read(codec, RT5640_I2S1_SDP); + val = (val & RT5640_I2S_IF_MASK) >> RT5640_I2S_IF_SFT; + switch (dai_id) { + case RT5640_AIF1: + switch (val) { + case RT5640_IF_123: + case RT5640_IF_132: + ret |= RT5640_U_IF1; + break; + case RT5640_IF_113: + ret |= RT5640_U_IF1; + case RT5640_IF_312: + case RT5640_IF_213: + ret |= RT5640_U_IF2; + break; + } + break; + + case RT5640_AIF2: + switch (val) { + case RT5640_IF_231: + case RT5640_IF_213: + ret |= RT5640_U_IF1; + break; + case RT5640_IF_223: + ret |= RT5640_U_IF1; + case RT5640_IF_123: + case RT5640_IF_321: + ret |= RT5640_U_IF2; + break; + } + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5640_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk, dai_sel; + int pre_div, bclk_ms, frame_size; + + rt5640->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return frame_size; + } + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + rt5640->bclk[dai->id] = rt5640->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5640->bclk[dai->id], rt5640->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5640_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5640_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5640_I2S_DL_8; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5640_U_IF1) { + mask_clk = RT5640_I2S_BCLK_MS1_MASK | RT5640_I2S_PD1_MASK; + val_clk = bclk_ms << RT5640_I2S_BCLK_MS1_SFT | + pre_div << RT5640_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5640_I2S1_SDP, + RT5640_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5640_ADDA_CLK1, mask_clk, val_clk); + } + if (dai_sel & RT5640_U_IF2) { + mask_clk = RT5640_I2S_BCLK_MS2_MASK | RT5640_I2S_PD2_MASK; + val_clk = bclk_ms << RT5640_I2S_BCLK_MS2_SFT | + pre_div << RT5640_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5640_I2S2_SDP, + RT5640_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5640_ADDA_CLK1, mask_clk, val_clk); + } + + return 0; +} + +static int rt5640_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0, dai_sel; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5640->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5640_I2S_MS_S; + rt5640->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5640_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5640_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5640_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5640_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5640_U_IF1) { + snd_soc_update_bits(codec, RT5640_I2S1_SDP, + RT5640_I2S_MS_MASK | RT5640_I2S_BP_MASK | + RT5640_I2S_DF_MASK, reg_val); + } + if (dai_sel & RT5640_U_IF2) { + snd_soc_update_bits(codec, RT5640_I2S2_SDP, + RT5640_I2S_MS_MASK | RT5640_I2S_BP_MASK | + RT5640_I2S_DF_MASK, reg_val); + } + + return 0; +} + +static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5640->sysclk && clk_id == rt5640->sysclk_src) + return 0; + + switch (clk_id) { + case RT5640_SCLK_S_MCLK: + reg_val |= RT5640_SCLK_SRC_MCLK; + break; + case RT5640_SCLK_S_PLL1: + reg_val |= RT5640_SCLK_SRC_PLL1; + break; + case RT5640_SCLK_S_PLL1_TK: + reg_val |= RT5640_SCLK_SRC_PLL1T; + break; + case RT5640_SCLK_S_RCCLK: + reg_val |= RT5640_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, reg_val); + rt5640->sysclk = freq; + rt5640->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + return 0; +} + +/** + * rt5640_pll_calc - Calculate PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calculate M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5640_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5640_pll_code *pll_code) +{ + int max_n = RT5640_PLL_N_MAX, max_m = RT5640_PLL_M_MAX; + int n = 0, m = 0, red, n_t, m_t, in_t, out_t; + int red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5640_PLL_INP_MAX < freq_in || RT5640_PLL_INP_MIN > freq_in) + return -EINVAL; + + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = (freq_in >> 1) + (freq_in >> 2) * n_t; + if (in_t < 0) + continue; + if (in_t == freq_out) { + bypass = true; + n = n_t; + goto code_find; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - freq_out); + if (red < red_t) { + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = 2; + return 0; +} + +static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + struct rt5640_pll_code *pll_code = &rt5640->pll_code; + int ret, dai_sel; + + if (source == rt5640->pll_src && freq_in == rt5640->pll_in && + freq_out == rt5640->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5640->pll_in = 0; + rt5640->pll_out = 0; + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, RT5640_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5640_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_MCLK); + break; + case RT5640_PLL1_S_BCLK1: + case RT5640_PLL1_S_BCLK2: + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, + "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5640_U_IF1) { + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); + } + if (dai_sel & RT5640_U_IF2) { + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5640_pll_calc(freq_in, freq_out, pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=2\n", pll_code->m_bp, + (pll_code->m_bp ? 0 : pll_code->m_code), pll_code->n_code); + + snd_soc_write(codec, RT5640_PLL_CTRL1, + pll_code->n_code << RT5640_PLL_N_SFT | pll_code->k_code); + snd_soc_write(codec, RT5640_PLL_CTRL2, + (pll_code->m_bp ? 0 : pll_code->m_code) << RT5640_PLL_M_SFT | + pll_code->m_bp << RT5640_PLL_M_BP_SFT); + + rt5640->pll_in = freq_in; + rt5640->pll_out = freq_out; + rt5640->pll_src = source; + + return 0; +} + +static int rt5640_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + regcache_cache_only(rt5640->regmap, false); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_VREF1 | RT5640_PWR_MB | + RT5640_PWR_BG | RT5640_PWR_VREF2, + RT5640_PWR_VREF1 | RT5640_PWR_MB | + RT5640_PWR_BG | RT5640_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, + RT5640_PWR_FV1 | RT5640_PWR_FV2); + regcache_sync(rt5640->regmap); + snd_soc_update_bits(codec, RT5640_DUMMY1, + 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5640_DEPOP_M1, + 0x001d, 0x0019); + snd_soc_update_bits(codec, RT5640_DEPOP_M2, + 0x2000, 0x2000); + snd_soc_update_bits(codec, RT5640_MICBIAS, + 0x0030, 0x0030); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5640_DEPOP_M1, 0x0004); + snd_soc_write(codec, RT5640_DEPOP_M2, 0x1100); + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x1, 0); + snd_soc_write(codec, RT5640_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5640_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5640_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5640_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5640_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5640_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5640_probe(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5640->codec = codec; + codec->control_data = rt5640->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + codec->dapm.idle_bias_off = 1; + rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); + snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); + snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); + snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + + return 0; +} + +static int rt5640_remove(struct snd_soc_codec *codec) +{ + rt5640_reset(codec); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5640_suspend(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); + rt5640_reset(codec); + regcache_cache_only(rt5640->regmap, true); + regcache_mark_dirty(rt5640->regmap); + + return 0; +} + +static int rt5640_resume(struct snd_soc_codec *codec) +{ + rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define rt5640_suspend NULL +#define rt5640_resume NULL +#endif + +#define RT5640_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5640_aif_dai_ops = { + .hw_params = rt5640_hw_params, + .set_fmt = rt5640_set_dai_fmt, + .set_sysclk = rt5640_set_dai_sysclk, + .set_pll = rt5640_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5640_dai[] = { + { + .name = "rt5640-aif1", + .id = RT5640_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, + { + .name = "rt5640-aif2", + .id = RT5640_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { + .probe = rt5640_probe, + .remove = rt5640_remove, + .suspend = rt5640_suspend, + .resume = rt5640_resume, + .set_bias_level = rt5640_set_bias_level, + .controls = rt5640_snd_controls, + .num_controls = ARRAY_SIZE(rt5640_snd_controls), + .dapm_widgets = rt5640_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5640_dapm_widgets), + .dapm_routes = rt5640_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5640_dapm_routes), +}; + +static const struct regmap_config rt5640_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) * + RT5640_PR_SPACING), + .volatile_reg = rt5640_volatile_register, + .readable_reg = rt5640_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5640_reg, + .num_reg_defaults = ARRAY_SIZE(rt5640_reg), + .ranges = rt5640_ranges, + .num_ranges = ARRAY_SIZE(rt5640_ranges), +}; + +static const struct i2c_device_id rt5640_i2c_id[] = { + { "rt5640", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); + +static int rt5640_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5640_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5640_priv *rt5640; + int ret; + unsigned int val; + + rt5640 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5640_priv), + GFP_KERNEL); + if (NULL == rt5640) + return -ENOMEM; + + rt5640->regmap = devm_regmap_init_i2c(i2c, &rt5640_regmap); + if (IS_ERR(rt5640->regmap)) { + ret = PTR_ERR(rt5640->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + if (pdata) + rt5640->pdata = *pdata; + + i2c_set_clientdata(i2c, rt5640); + + if (rt5640->pdata.ldo1_en) { + ret = devm_gpio_request_one(&i2c->dev, rt5640->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, + "RT5640 LDO1_EN"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDO1_EN %d: %d\n", + rt5640->pdata.ldo1_en, ret); + return ret; + } + msleep(400); + } + + regmap_read(rt5640->regmap, RT5640_VENDOR_ID2, &val); + if ((val != RT5640_DEVICE_ID)) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5640/39\n", val); + return -ENODEV; + } + + regmap_write(rt5640->regmap, RT5640_RESET, 0); + + ret = regmap_register_patch(rt5640->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5640->pdata.in1_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, + RT5640_IN_DF1, RT5640_IN_DF1); + + if (rt5640->pdata.in2_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, + RT5640_IN_DF2, RT5640_IN_DF2); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5640_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt5640_i2c_driver = { + .driver = { + .name = "rt5640", + .owner = THIS_MODULE, + }, + .probe = rt5640_i2c_probe, + .remove = rt5640_i2c_remove, + .id_table = rt5640_i2c_id, +}; +module_i2c_driver(rt5640_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5640 driver"); +MODULE_AUTHOR("Johnny Hsu "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h new file mode 100644 index 000000000000..c48286d7118f --- /dev/null +++ b/sound/soc/codecs/rt5640.h @@ -0,0 +1,2092 @@ +/* + * rt5640.h -- RT5640 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _RT5640_H +#define _RT5640_H + +#include + +/* Info */ +#define RT5640_RESET 0x00 +#define RT5640_VENDOR_ID 0xfd +#define RT5640_VENDOR_ID1 0xfe +#define RT5640_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5640_SPK_VOL 0x01 +#define RT5640_HP_VOL 0x02 +#define RT5640_OUTPUT 0x03 +#define RT5640_MONO_OUT 0x04 +/* I/O - Input */ +#define RT5640_IN1_IN2 0x0d +#define RT5640_IN3_IN4 0x0e +#define RT5640_INL_INR_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5640_DAC1_DIG_VOL 0x19 +#define RT5640_DAC2_DIG_VOL 0x1a +#define RT5640_DAC2_CTRL 0x1b +#define RT5640_ADC_DIG_VOL 0x1c +#define RT5640_ADC_DATA 0x1d +#define RT5640_ADC_BST_VOL 0x1e +/* Mixer - D-D */ +#define RT5640_STO_ADC_MIXER 0x27 +#define RT5640_MONO_ADC_MIXER 0x28 +#define RT5640_AD_DA_MIXER 0x29 +#define RT5640_STO_DAC_MIXER 0x2a +#define RT5640_MONO_DAC_MIXER 0x2b +#define RT5640_DIG_MIXER 0x2c +#define RT5640_DSP_PATH1 0x2d +#define RT5640_DSP_PATH2 0x2e +#define RT5640_DIG_INF_DATA 0x2f +/* Mixer - ADC */ +#define RT5640_REC_L1_MIXER 0x3b +#define RT5640_REC_L2_MIXER 0x3c +#define RT5640_REC_R1_MIXER 0x3d +#define RT5640_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5640_HPO_MIXER 0x45 +#define RT5640_SPK_L_MIXER 0x46 +#define RT5640_SPK_R_MIXER 0x47 +#define RT5640_SPO_L_MIXER 0x48 +#define RT5640_SPO_R_MIXER 0x49 +#define RT5640_SPO_CLSD_RATIO 0x4a +#define RT5640_MONO_MIXER 0x4c +#define RT5640_OUT_L1_MIXER 0x4d +#define RT5640_OUT_L2_MIXER 0x4e +#define RT5640_OUT_L3_MIXER 0x4f +#define RT5640_OUT_R1_MIXER 0x50 +#define RT5640_OUT_R2_MIXER 0x51 +#define RT5640_OUT_R3_MIXER 0x52 +#define RT5640_LOUT_MIXER 0x53 +/* Power */ +#define RT5640_PWR_DIG1 0x61 +#define RT5640_PWR_DIG2 0x62 +#define RT5640_PWR_ANLG1 0x63 +#define RT5640_PWR_ANLG2 0x64 +#define RT5640_PWR_MIXER 0x65 +#define RT5640_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5640_PRIV_INDEX 0x6a +#define RT5640_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5640_I2S1_SDP 0x70 +#define RT5640_I2S2_SDP 0x71 +#define RT5640_ADDA_CLK1 0x73 +#define RT5640_ADDA_CLK2 0x74 +#define RT5640_DMIC 0x75 +/* Function - Analog */ +#define RT5640_GLB_CLK 0x80 +#define RT5640_PLL_CTRL1 0x81 +#define RT5640_PLL_CTRL2 0x82 +#define RT5640_ASRC_1 0x83 +#define RT5640_ASRC_2 0x84 +#define RT5640_ASRC_3 0x85 +#define RT5640_ASRC_4 0x89 +#define RT5640_ASRC_5 0x8a +#define RT5640_HP_OVCD 0x8b +#define RT5640_CLS_D_OVCD 0x8c +#define RT5640_CLS_D_OUT 0x8d +#define RT5640_DEPOP_M1 0x8e +#define RT5640_DEPOP_M2 0x8f +#define RT5640_DEPOP_M3 0x90 +#define RT5640_CHARGE_PUMP 0x91 +#define RT5640_PV_DET_SPK_G 0x92 +#define RT5640_MICBIAS 0x93 +/* Function - Digital */ +#define RT5640_EQ_CTRL1 0xb0 +#define RT5640_EQ_CTRL2 0xb1 +#define RT5640_WIND_FILTER 0xb2 +#define RT5640_DRC_AGC_1 0xb4 +#define RT5640_DRC_AGC_2 0xb5 +#define RT5640_DRC_AGC_3 0xb6 +#define RT5640_SVOL_ZC 0xb7 +#define RT5640_ANC_CTRL1 0xb8 +#define RT5640_ANC_CTRL2 0xb9 +#define RT5640_ANC_CTRL3 0xba +#define RT5640_JD_CTRL 0xbb +#define RT5640_ANC_JD 0xbc +#define RT5640_IRQ_CTRL1 0xbd +#define RT5640_IRQ_CTRL2 0xbe +#define RT5640_INT_IRQ_ST 0xbf +#define RT5640_GPIO_CTRL1 0xc0 +#define RT5640_GPIO_CTRL2 0xc1 +#define RT5640_GPIO_CTRL3 0xc2 +#define RT5640_DSP_CTRL1 0xc4 +#define RT5640_DSP_CTRL2 0xc5 +#define RT5640_DSP_CTRL3 0xc6 +#define RT5640_DSP_CTRL4 0xc7 +#define RT5640_PGM_REG_ARR1 0xc8 +#define RT5640_PGM_REG_ARR2 0xc9 +#define RT5640_PGM_REG_ARR3 0xca +#define RT5640_PGM_REG_ARR4 0xcb +#define RT5640_PGM_REG_ARR5 0xcc +#define RT5640_SCB_FUNC 0xcd +#define RT5640_SCB_CTRL 0xce +#define RT5640_BASE_BACK 0xcf +#define RT5640_MP3_PLUS1 0xd0 +#define RT5640_MP3_PLUS2 0xd1 +#define RT5640_3D_HP 0xd2 +#define RT5640_ADJ_HPF 0xd3 +#define RT5640_HP_CALIB_AMP_DET 0xd6 +#define RT5640_HP_CALIB2 0xd7 +#define RT5640_SV_ZCD1 0xd9 +#define RT5640_SV_ZCD2 0xda +/* Dummy Register */ +#define RT5640_DUMMY1 0xfa +#define RT5640_DUMMY2 0xfb +#define RT5640_DUMMY3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5640_3D_SPK 0x63 +#define RT5640_WND_1 0x6c +#define RT5640_WND_2 0x6d +#define RT5640_WND_3 0x6e +#define RT5640_WND_4 0x6f +#define RT5640_WND_5 0x70 +#define RT5640_WND_8 0x73 +#define RT5640_DIP_SPK_INF 0x75 +#define RT5640_EQ_BW_LOP 0xa0 +#define RT5640_EQ_GN_LOP 0xa1 +#define RT5640_EQ_FC_BP1 0xa2 +#define RT5640_EQ_BW_BP1 0xa3 +#define RT5640_EQ_GN_BP1 0xa4 +#define RT5640_EQ_FC_BP2 0xa5 +#define RT5640_EQ_BW_BP2 0xa6 +#define RT5640_EQ_GN_BP2 0xa7 +#define RT5640_EQ_FC_BP3 0xa8 +#define RT5640_EQ_BW_BP3 0xa9 +#define RT5640_EQ_GN_BP3 0xaa +#define RT5640_EQ_FC_BP4 0xab +#define RT5640_EQ_BW_BP4 0xac +#define RT5640_EQ_GN_BP4 0xad +#define RT5640_EQ_FC_HIP1 0xae +#define RT5640_EQ_GN_HIP1 0xaf +#define RT5640_EQ_FC_HIP2 0xb0 +#define RT5640_EQ_BW_HIP2 0xb1 +#define RT5640_EQ_GN_HIP2 0xb2 +#define RT5640_EQ_PRE_VOL 0xb3 +#define RT5640_EQ_PST_VOL 0xb4 + +/* global definition */ +#define RT5640_L_MUTE (0x1 << 15) +#define RT5640_L_MUTE_SFT 15 +#define RT5640_VOL_L_MUTE (0x1 << 14) +#define RT5640_VOL_L_SFT 14 +#define RT5640_R_MUTE (0x1 << 7) +#define RT5640_R_MUTE_SFT 7 +#define RT5640_VOL_R_MUTE (0x1 << 6) +#define RT5640_VOL_R_SFT 6 +#define RT5640_L_VOL_MASK (0x3f << 8) +#define RT5640_L_VOL_SFT 8 +#define RT5640_R_VOL_MASK (0x3f) +#define RT5640_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +/* IN3 and IN4 Control (0x0e) */ +#define RT5640_BST_SFT1 12 +#define RT5640_BST_SFT2 8 +#define RT5640_IN_DF1 (0x1 << 7) +#define RT5640_IN_SFT1 7 +#define RT5640_IN_DF2 (0x1 << 6) +#define RT5640_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5640_INL_SEL_MASK (0x1 << 15) +#define RT5640_INL_SEL_SFT 15 +#define RT5640_INL_SEL_IN4P (0x0 << 15) +#define RT5640_INL_SEL_MONOP (0x1 << 15) +#define RT5640_INL_VOL_MASK (0x1f << 8) +#define RT5640_INL_VOL_SFT 8 +#define RT5640_INR_SEL_MASK (0x1 << 7) +#define RT5640_INR_SEL_SFT 7 +#define RT5640_INR_SEL_IN4N (0x0 << 7) +#define RT5640_INR_SEL_MONON (0x1 << 7) +#define RT5640_INR_VOL_MASK (0x1f) +#define RT5640_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5640_DAC_L1_VOL_MASK (0xff << 8) +#define RT5640_DAC_L1_VOL_SFT 8 +#define RT5640_DAC_R1_VOL_MASK (0xff) +#define RT5640_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5640_DAC_L2_VOL_MASK (0xff << 8) +#define RT5640_DAC_L2_VOL_SFT 8 +#define RT5640_DAC_R2_VOL_MASK (0xff) +#define RT5640_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5640_M_DAC_L2_VOL (0x1 << 13) +#define RT5640_M_DAC_L2_VOL_SFT 13 +#define RT5640_M_DAC_R2_VOL (0x1 << 12) +#define RT5640_M_DAC_R2_VOL_SFT 12 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5640_ADC_L_VOL_MASK (0x7f << 8) +#define RT5640_ADC_L_VOL_SFT 8 +#define RT5640_ADC_R_VOL_MASK (0x7f) +#define RT5640_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5640_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5640_MONO_ADC_L_VOL_SFT 8 +#define RT5640_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5640_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5640_ADC_L_BST_MASK (0x3 << 14) +#define RT5640_ADC_L_BST_SFT 14 +#define RT5640_ADC_R_BST_MASK (0x3 << 12) +#define RT5640_ADC_R_BST_SFT 12 +#define RT5640_ADC_COMP_MASK (0x3 << 10) +#define RT5640_ADC_COMP_SFT 10 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5640_M_ADC_L1 (0x1 << 14) +#define RT5640_M_ADC_L1_SFT 14 +#define RT5640_M_ADC_L2 (0x1 << 13) +#define RT5640_M_ADC_L2_SFT 13 +#define RT5640_ADC_1_SRC_MASK (0x1 << 12) +#define RT5640_ADC_1_SRC_SFT 12 +#define RT5640_ADC_1_SRC_ADC (0x1 << 12) +#define RT5640_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5640_ADC_2_SRC_MASK (0x3 << 10) +#define RT5640_ADC_2_SRC_SFT 10 +#define RT5640_ADC_2_SRC_DMIC1 (0x0 << 10) +#define RT5640_ADC_2_SRC_DMIC2 (0x1 << 10) +#define RT5640_ADC_2_SRC_DACMIX (0x2 << 10) +#define RT5640_M_ADC_R1 (0x1 << 6) +#define RT5640_M_ADC_R1_SFT 6 +#define RT5640_M_ADC_R2 (0x1 << 5) +#define RT5640_M_ADC_R2_SFT 5 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5640_M_MONO_ADC_L1 (0x1 << 14) +#define RT5640_M_MONO_ADC_L1_SFT 14 +#define RT5640_M_MONO_ADC_L2 (0x1 << 13) +#define RT5640_M_MONO_ADC_L2_SFT 13 +#define RT5640_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5640_MONO_ADC_L1_SRC_SFT 12 +#define RT5640_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5640_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5640_MONO_ADC_L2_SRC_MASK (0x3 << 10) +#define RT5640_MONO_ADC_L2_SRC_SFT 10 +#define RT5640_MONO_ADC_L2_SRC_DMIC_L1 (0x0 << 10) +#define RT5640_MONO_ADC_L2_SRC_DMIC_L2 (0x1 << 10) +#define RT5640_MONO_ADC_L2_SRC_DACMIXL (0x2 << 10) +#define RT5640_M_MONO_ADC_R1 (0x1 << 6) +#define RT5640_M_MONO_ADC_R1_SFT 6 +#define RT5640_M_MONO_ADC_R2 (0x1 << 5) +#define RT5640_M_MONO_ADC_R2_SFT 5 +#define RT5640_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5640_MONO_ADC_R1_SRC_SFT 4 +#define RT5640_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5640_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5640_MONO_ADC_R2_SRC_MASK (0x3 << 2) +#define RT5640_MONO_ADC_R2_SRC_SFT 2 +#define RT5640_MONO_ADC_R2_SRC_DMIC_R1 (0x0 << 2) +#define RT5640_MONO_ADC_R2_SRC_DMIC_R2 (0x1 << 2) +#define RT5640_MONO_ADC_R2_SRC_DACMIXR (0x2 << 2) + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5640_M_ADCMIX_L (0x1 << 15) +#define RT5640_M_ADCMIX_L_SFT 15 +#define RT5640_M_IF1_DAC_L (0x1 << 14) +#define RT5640_M_IF1_DAC_L_SFT 14 +#define RT5640_M_ADCMIX_R (0x1 << 7) +#define RT5640_M_ADCMIX_R_SFT 7 +#define RT5640_M_IF1_DAC_R (0x1 << 6) +#define RT5640_M_IF1_DAC_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5640_M_DAC_L1 (0x1 << 14) +#define RT5640_M_DAC_L1_SFT 14 +#define RT5640_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5640_DAC_L1_STO_L_VOL_SFT 13 +#define RT5640_M_DAC_L2 (0x1 << 12) +#define RT5640_M_DAC_L2_SFT 12 +#define RT5640_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5640_DAC_L2_STO_L_VOL_SFT 11 +#define RT5640_M_ANC_DAC_L (0x1 << 10) +#define RT5640_M_ANC_DAC_L_SFT 10 +#define RT5640_M_DAC_R1 (0x1 << 6) +#define RT5640_M_DAC_R1_SFT 6 +#define RT5640_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5640_DAC_R1_STO_R_VOL_SFT 5 +#define RT5640_M_DAC_R2 (0x1 << 4) +#define RT5640_M_DAC_R2_SFT 4 +#define RT5640_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5640_DAC_R2_STO_R_VOL_SFT 3 +#define RT5640_M_ANC_DAC_R (0x1 << 2) +#define RT5640_M_ANC_DAC_R_SFT 2 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5640_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5640_M_DAC_L1_MONO_L_SFT 14 +#define RT5640_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5640_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5640_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5640_M_DAC_L2_MONO_L_SFT 12 +#define RT5640_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5640_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5640_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5640_M_DAC_R2_MONO_L_SFT 10 +#define RT5640_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5640_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5640_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5640_M_DAC_R1_MONO_R_SFT 6 +#define RT5640_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5640_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5640_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5640_M_DAC_R2_MONO_R_SFT 4 +#define RT5640_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5640_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5640_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5640_M_DAC_L2_MONO_R_SFT 2 +#define RT5640_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5640_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5640_M_STO_L_DAC_L (0x1 << 15) +#define RT5640_M_STO_L_DAC_L_SFT 15 +#define RT5640_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5640_STO_L_DAC_L_VOL_SFT 14 +#define RT5640_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5640_M_DAC_L2_DAC_L_SFT 13 +#define RT5640_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5640_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5640_M_STO_R_DAC_R (0x1 << 11) +#define RT5640_M_STO_R_DAC_R_SFT 11 +#define RT5640_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5640_STO_R_DAC_R_VOL_SFT 10 +#define RT5640_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5640_M_DAC_R2_DAC_R_SFT 9 +#define RT5640_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5640_DAC_R2_DAC_R_VOL_SFT 8 + +/* DSP Path Control 1 (0x2d) */ +#define RT5640_RXDP_SRC_MASK (0x1 << 15) +#define RT5640_RXDP_SRC_SFT 15 +#define RT5640_RXDP_SRC_NOR (0x0 << 15) +#define RT5640_RXDP_SRC_DIV3 (0x1 << 15) +#define RT5640_TXDP_SRC_MASK (0x1 << 14) +#define RT5640_TXDP_SRC_SFT 14 +#define RT5640_TXDP_SRC_NOR (0x0 << 14) +#define RT5640_TXDP_SRC_DIV3 (0x1 << 14) + +/* DSP Path Control 2 (0x2e) */ +#define RT5640_DAC_L2_SEL_MASK (0x3 << 14) +#define RT5640_DAC_L2_SEL_SFT 14 +#define RT5640_DAC_L2_SEL_IF2 (0x0 << 14) +#define RT5640_DAC_L2_SEL_IF3 (0x1 << 14) +#define RT5640_DAC_L2_SEL_TXDC (0x2 << 14) +#define RT5640_DAC_L2_SEL_BASS (0x3 << 14) +#define RT5640_DAC_R2_SEL_MASK (0x3 << 12) +#define RT5640_DAC_R2_SEL_SFT 12 +#define RT5640_DAC_R2_SEL_IF2 (0x0 << 12) +#define RT5640_DAC_R2_SEL_IF3 (0x1 << 12) +#define RT5640_DAC_R2_SEL_TXDC (0x2 << 12) +#define RT5640_IF2_ADC_L_SEL_MASK (0x1 << 11) +#define RT5640_IF2_ADC_L_SEL_SFT 11 +#define RT5640_IF2_ADC_L_SEL_TXDP (0x0 << 11) +#define RT5640_IF2_ADC_L_SEL_PASS (0x1 << 11) +#define RT5640_IF2_ADC_R_SEL_MASK (0x1 << 10) +#define RT5640_IF2_ADC_R_SEL_SFT 10 +#define RT5640_IF2_ADC_R_SEL_TXDP (0x0 << 10) +#define RT5640_IF2_ADC_R_SEL_PASS (0x1 << 10) +#define RT5640_RXDC_SEL_MASK (0x3 << 8) +#define RT5640_RXDC_SEL_SFT 8 +#define RT5640_RXDC_SEL_NOR (0x0 << 8) +#define RT5640_RXDC_SEL_L2R (0x1 << 8) +#define RT5640_RXDC_SEL_R2L (0x2 << 8) +#define RT5640_RXDC_SEL_SWAP (0x3 << 8) +#define RT5640_RXDP_SEL_MASK (0x3 << 6) +#define RT5640_RXDP_SEL_SFT 6 +#define RT5640_RXDP_SEL_NOR (0x0 << 6) +#define RT5640_RXDP_SEL_L2R (0x1 << 6) +#define RT5640_RXDP_SEL_R2L (0x2 << 6) +#define RT5640_RXDP_SEL_SWAP (0x3 << 6) +#define RT5640_TXDC_SEL_MASK (0x3 << 4) +#define RT5640_TXDC_SEL_SFT 4 +#define RT5640_TXDC_SEL_NOR (0x0 << 4) +#define RT5640_TXDC_SEL_L2R (0x1 << 4) +#define RT5640_TXDC_SEL_R2L (0x2 << 4) +#define RT5640_TXDC_SEL_SWAP (0x3 << 4) +#define RT5640_TXDP_SEL_MASK (0x3 << 2) +#define RT5640_TXDP_SEL_SFT 2 +#define RT5640_TXDP_SEL_NOR (0x0 << 2) +#define RT5640_TXDP_SEL_L2R (0x1 << 2) +#define RT5640_TXDP_SEL_R2L (0x2 << 2) +#define RT5640_TRXDP_SEL_SWAP (0x3 << 2) + +/* Digital Interface Data Control (0x2f) */ +#define RT5640_IF1_DAC_SEL_MASK (0x3 << 14) +#define RT5640_IF1_DAC_SEL_SFT 14 +#define RT5640_IF1_DAC_SEL_NOR (0x0 << 14) +#define RT5640_IF1_DAC_SEL_L2R (0x1 << 14) +#define RT5640_IF1_DAC_SEL_R2L (0x2 << 14) +#define RT5640_IF1_DAC_SEL_SWAP (0x3 << 14) +#define RT5640_IF1_ADC_SEL_MASK (0x3 << 12) +#define RT5640_IF1_ADC_SEL_SFT 12 +#define RT5640_IF1_ADC_SEL_NOR (0x0 << 12) +#define RT5640_IF1_ADC_SEL_L2R (0x1 << 12) +#define RT5640_IF1_ADC_SEL_R2L (0x2 << 12) +#define RT5640_IF1_ADC_SEL_SWAP (0x3 << 12) +#define RT5640_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5640_IF2_DAC_SEL_SFT 10 +#define RT5640_IF2_DAC_SEL_NOR (0x0 << 10) +#define RT5640_IF2_DAC_SEL_L2R (0x1 << 10) +#define RT5640_IF2_DAC_SEL_R2L (0x2 << 10) +#define RT5640_IF2_DAC_SEL_SWAP (0x3 << 10) +#define RT5640_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5640_IF2_ADC_SEL_SFT 8 +#define RT5640_IF2_ADC_SEL_NOR (0x0 << 8) +#define RT5640_IF2_ADC_SEL_L2R (0x1 << 8) +#define RT5640_IF2_ADC_SEL_R2L (0x2 << 8) +#define RT5640_IF2_ADC_SEL_SWAP (0x3 << 8) +#define RT5640_IF3_DAC_SEL_MASK (0x3 << 6) +#define RT5640_IF3_DAC_SEL_SFT 6 +#define RT5640_IF3_DAC_SEL_NOR (0x0 << 6) +#define RT5640_IF3_DAC_SEL_L2R (0x1 << 6) +#define RT5640_IF3_DAC_SEL_R2L (0x2 << 6) +#define RT5640_IF3_DAC_SEL_SWAP (0x3 << 6) +#define RT5640_IF3_ADC_SEL_MASK (0x3 << 4) +#define RT5640_IF3_ADC_SEL_SFT 4 +#define RT5640_IF3_ADC_SEL_NOR (0x0 << 4) +#define RT5640_IF3_ADC_SEL_L2R (0x1 << 4) +#define RT5640_IF3_ADC_SEL_R2L (0x2 << 4) +#define RT5640_IF3_ADC_SEL_SWAP (0x3 << 4) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5640_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5640_G_HP_L_RM_L_SFT 13 +#define RT5640_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5640_G_IN_L_RM_L_SFT 10 +#define RT5640_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5640_G_BST4_RM_L_SFT 7 +#define RT5640_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5640_G_BST3_RM_L_SFT 4 +#define RT5640_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5640_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5640_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5640_G_BST1_RM_L_SFT 13 +#define RT5640_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5640_G_OM_L_RM_L_SFT 10 +#define RT5640_M_HP_L_RM_L (0x1 << 6) +#define RT5640_M_HP_L_RM_L_SFT 6 +#define RT5640_M_IN_L_RM_L (0x1 << 5) +#define RT5640_M_IN_L_RM_L_SFT 5 +#define RT5640_M_BST4_RM_L (0x1 << 4) +#define RT5640_M_BST4_RM_L_SFT 4 +#define RT5640_M_BST3_RM_L (0x1 << 3) +#define RT5640_M_BST3_RM_L_SFT 3 +#define RT5640_M_BST2_RM_L (0x1 << 2) +#define RT5640_M_BST2_RM_L_SFT 2 +#define RT5640_M_BST1_RM_L (0x1 << 1) +#define RT5640_M_BST1_RM_L_SFT 1 +#define RT5640_M_OM_L_RM_L (0x1) +#define RT5640_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5640_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5640_G_HP_R_RM_R_SFT 13 +#define RT5640_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5640_G_IN_R_RM_R_SFT 10 +#define RT5640_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5640_G_BST4_RM_R_SFT 7 +#define RT5640_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5640_G_BST3_RM_R_SFT 4 +#define RT5640_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5640_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5640_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5640_G_BST1_RM_R_SFT 13 +#define RT5640_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5640_G_OM_R_RM_R_SFT 10 +#define RT5640_M_HP_R_RM_R (0x1 << 6) +#define RT5640_M_HP_R_RM_R_SFT 6 +#define RT5640_M_IN_R_RM_R (0x1 << 5) +#define RT5640_M_IN_R_RM_R_SFT 5 +#define RT5640_M_BST4_RM_R (0x1 << 4) +#define RT5640_M_BST4_RM_R_SFT 4 +#define RT5640_M_BST3_RM_R (0x1 << 3) +#define RT5640_M_BST3_RM_R_SFT 3 +#define RT5640_M_BST2_RM_R (0x1 << 2) +#define RT5640_M_BST2_RM_R_SFT 2 +#define RT5640_M_BST1_RM_R (0x1 << 1) +#define RT5640_M_BST1_RM_R_SFT 1 +#define RT5640_M_OM_R_RM_R (0x1) +#define RT5640_M_OM_R_RM_R_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5640_M_DAC2_HM (0x1 << 15) +#define RT5640_M_DAC2_HM_SFT 15 +#define RT5640_M_DAC1_HM (0x1 << 14) +#define RT5640_M_DAC1_HM_SFT 14 +#define RT5640_M_HPVOL_HM (0x1 << 13) +#define RT5640_M_HPVOL_HM_SFT 13 +#define RT5640_G_HPOMIX_MASK (0x1 << 12) +#define RT5640_G_HPOMIX_SFT 12 + +/* SPK Left Mixer Control (0x46) */ +#define RT5640_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5640_G_RM_L_SM_L_SFT 14 +#define RT5640_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5640_G_IN_L_SM_L_SFT 12 +#define RT5640_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5640_G_DAC_L1_SM_L_SFT 10 +#define RT5640_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5640_G_DAC_L2_SM_L_SFT 8 +#define RT5640_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5640_G_OM_L_SM_L_SFT 6 +#define RT5640_M_RM_L_SM_L (0x1 << 5) +#define RT5640_M_RM_L_SM_L_SFT 5 +#define RT5640_M_IN_L_SM_L (0x1 << 4) +#define RT5640_M_IN_L_SM_L_SFT 4 +#define RT5640_M_DAC_L1_SM_L (0x1 << 3) +#define RT5640_M_DAC_L1_SM_L_SFT 3 +#define RT5640_M_DAC_L2_SM_L (0x1 << 2) +#define RT5640_M_DAC_L2_SM_L_SFT 2 +#define RT5640_M_OM_L_SM_L (0x1 << 1) +#define RT5640_M_OM_L_SM_L_SFT 1 + +/* SPK Right Mixer Control (0x47) */ +#define RT5640_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5640_G_RM_R_SM_R_SFT 14 +#define RT5640_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5640_G_IN_R_SM_R_SFT 12 +#define RT5640_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5640_G_DAC_R1_SM_R_SFT 10 +#define RT5640_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5640_G_DAC_R2_SM_R_SFT 8 +#define RT5640_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5640_G_OM_R_SM_R_SFT 6 +#define RT5640_M_RM_R_SM_R (0x1 << 5) +#define RT5640_M_RM_R_SM_R_SFT 5 +#define RT5640_M_IN_R_SM_R (0x1 << 4) +#define RT5640_M_IN_R_SM_R_SFT 4 +#define RT5640_M_DAC_R1_SM_R (0x1 << 3) +#define RT5640_M_DAC_R1_SM_R_SFT 3 +#define RT5640_M_DAC_R2_SM_R (0x1 << 2) +#define RT5640_M_DAC_R2_SM_R_SFT 2 +#define RT5640_M_OM_R_SM_R (0x1 << 1) +#define RT5640_M_OM_R_SM_R_SFT 1 + +/* SPOLMIX Control (0x48) */ +#define RT5640_M_DAC_R1_SPM_L (0x1 << 15) +#define RT5640_M_DAC_R1_SPM_L_SFT 15 +#define RT5640_M_DAC_L1_SPM_L (0x1 << 14) +#define RT5640_M_DAC_L1_SPM_L_SFT 14 +#define RT5640_M_SV_R_SPM_L (0x1 << 13) +#define RT5640_M_SV_R_SPM_L_SFT 13 +#define RT5640_M_SV_L_SPM_L (0x1 << 12) +#define RT5640_M_SV_L_SPM_L_SFT 12 +#define RT5640_M_BST1_SPM_L (0x1 << 11) +#define RT5640_M_BST1_SPM_L_SFT 11 + +/* SPORMIX Control (0x49) */ +#define RT5640_M_DAC_R1_SPM_R (0x1 << 13) +#define RT5640_M_DAC_R1_SPM_R_SFT 13 +#define RT5640_M_SV_R_SPM_R (0x1 << 12) +#define RT5640_M_SV_R_SPM_R_SFT 12 +#define RT5640_M_BST1_SPM_R (0x1 << 11) +#define RT5640_M_BST1_SPM_R_SFT 11 + +/* SPOLMIX / SPORMIX Ratio Control (0x4a) */ +#define RT5640_SPO_CLSD_RATIO_MASK (0x7) +#define RT5640_SPO_CLSD_RATIO_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5640_M_DAC_R2_MM (0x1 << 15) +#define RT5640_M_DAC_R2_MM_SFT 15 +#define RT5640_M_DAC_L2_MM (0x1 << 14) +#define RT5640_M_DAC_L2_MM_SFT 14 +#define RT5640_M_OV_R_MM (0x1 << 13) +#define RT5640_M_OV_R_MM_SFT 13 +#define RT5640_M_OV_L_MM (0x1 << 12) +#define RT5640_M_OV_L_MM_SFT 12 +#define RT5640_M_BST1_MM (0x1 << 11) +#define RT5640_M_BST1_MM_SFT 11 +#define RT5640_G_MONOMIX_MASK (0x1 << 10) +#define RT5640_G_MONOMIX_SFT 10 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5640_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5640_G_BST3_OM_L_SFT 13 +#define RT5640_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5640_G_BST2_OM_L_SFT 10 +#define RT5640_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5640_G_BST1_OM_L_SFT 7 +#define RT5640_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5640_G_IN_L_OM_L_SFT 4 +#define RT5640_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5640_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5640_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5640_G_DAC_R2_OM_L_SFT 13 +#define RT5640_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5640_G_DAC_L2_OM_L_SFT 10 +#define RT5640_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5640_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5640_M_SM_L_OM_L (0x1 << 8) +#define RT5640_M_SM_L_OM_L_SFT 8 +#define RT5640_M_BST3_OM_L (0x1 << 7) +#define RT5640_M_BST3_OM_L_SFT 7 +#define RT5640_M_BST2_OM_L (0x1 << 6) +#define RT5640_M_BST2_OM_L_SFT 6 +#define RT5640_M_BST1_OM_L (0x1 << 5) +#define RT5640_M_BST1_OM_L_SFT 5 +#define RT5640_M_IN_L_OM_L (0x1 << 4) +#define RT5640_M_IN_L_OM_L_SFT 4 +#define RT5640_M_RM_L_OM_L (0x1 << 3) +#define RT5640_M_RM_L_OM_L_SFT 3 +#define RT5640_M_DAC_R2_OM_L (0x1 << 2) +#define RT5640_M_DAC_R2_OM_L_SFT 2 +#define RT5640_M_DAC_L2_OM_L (0x1 << 1) +#define RT5640_M_DAC_L2_OM_L_SFT 1 +#define RT5640_M_DAC_L1_OM_L (0x1) +#define RT5640_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5640_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5640_G_BST4_OM_R_SFT 13 +#define RT5640_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5640_G_BST2_OM_R_SFT 10 +#define RT5640_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5640_G_BST1_OM_R_SFT 7 +#define RT5640_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5640_G_IN_R_OM_R_SFT 4 +#define RT5640_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5640_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5640_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5640_G_DAC_L2_OM_R_SFT 13 +#define RT5640_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5640_G_DAC_R2_OM_R_SFT 10 +#define RT5640_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5640_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5640_M_SM_L_OM_R (0x1 << 8) +#define RT5640_M_SM_L_OM_R_SFT 8 +#define RT5640_M_BST4_OM_R (0x1 << 7) +#define RT5640_M_BST4_OM_R_SFT 7 +#define RT5640_M_BST2_OM_R (0x1 << 6) +#define RT5640_M_BST2_OM_R_SFT 6 +#define RT5640_M_BST1_OM_R (0x1 << 5) +#define RT5640_M_BST1_OM_R_SFT 5 +#define RT5640_M_IN_R_OM_R (0x1 << 4) +#define RT5640_M_IN_R_OM_R_SFT 4 +#define RT5640_M_RM_R_OM_R (0x1 << 3) +#define RT5640_M_RM_R_OM_R_SFT 3 +#define RT5640_M_DAC_L2_OM_R (0x1 << 2) +#define RT5640_M_DAC_L2_OM_R_SFT 2 +#define RT5640_M_DAC_R2_OM_R (0x1 << 1) +#define RT5640_M_DAC_R2_OM_R_SFT 1 +#define RT5640_M_DAC_R1_OM_R (0x1) +#define RT5640_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5640_M_DAC_L1_LM (0x1 << 15) +#define RT5640_M_DAC_L1_LM_SFT 15 +#define RT5640_M_DAC_R1_LM (0x1 << 14) +#define RT5640_M_DAC_R1_LM_SFT 14 +#define RT5640_M_OV_L_LM (0x1 << 13) +#define RT5640_M_OV_L_LM_SFT 13 +#define RT5640_M_OV_R_LM (0x1 << 12) +#define RT5640_M_OV_R_LM_SFT 12 +#define RT5640_G_LOUTMIX_MASK (0x1 << 11) +#define RT5640_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5640_PWR_I2S1 (0x1 << 15) +#define RT5640_PWR_I2S1_BIT 15 +#define RT5640_PWR_I2S2 (0x1 << 14) +#define RT5640_PWR_I2S2_BIT 14 +#define RT5640_PWR_DAC_L1 (0x1 << 12) +#define RT5640_PWR_DAC_L1_BIT 12 +#define RT5640_PWR_DAC_R1 (0x1 << 11) +#define RT5640_PWR_DAC_R1_BIT 11 +#define RT5640_PWR_DAC_L2 (0x1 << 7) +#define RT5640_PWR_DAC_L2_BIT 7 +#define RT5640_PWR_DAC_R2 (0x1 << 6) +#define RT5640_PWR_DAC_R2_BIT 6 +#define RT5640_PWR_ADC_L (0x1 << 2) +#define RT5640_PWR_ADC_L_BIT 2 +#define RT5640_PWR_ADC_R (0x1 << 1) +#define RT5640_PWR_ADC_R_BIT 1 +#define RT5640_PWR_CLS_D (0x1) +#define RT5640_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5640_PWR_ADC_SF (0x1 << 15) +#define RT5640_PWR_ADC_SF_BIT 15 +#define RT5640_PWR_ADC_MF_L (0x1 << 14) +#define RT5640_PWR_ADC_MF_L_BIT 14 +#define RT5640_PWR_ADC_MF_R (0x1 << 13) +#define RT5640_PWR_ADC_MF_R_BIT 13 +#define RT5640_PWR_I2S_DSP (0x1 << 12) +#define RT5640_PWR_I2S_DSP_BIT 12 + +/* Power Management for Analog 1 (0x63) */ +#define RT5640_PWR_VREF1 (0x1 << 15) +#define RT5640_PWR_VREF1_BIT 15 +#define RT5640_PWR_FV1 (0x1 << 14) +#define RT5640_PWR_FV1_BIT 14 +#define RT5640_PWR_MB (0x1 << 13) +#define RT5640_PWR_MB_BIT 13 +#define RT5640_PWR_LM (0x1 << 12) +#define RT5640_PWR_LM_BIT 12 +#define RT5640_PWR_BG (0x1 << 11) +#define RT5640_PWR_BG_BIT 11 +#define RT5640_PWR_MM (0x1 << 10) +#define RT5640_PWR_MM_BIT 10 +#define RT5640_PWR_MA (0x1 << 8) +#define RT5640_PWR_MA_BIT 8 +#define RT5640_PWR_HP_L (0x1 << 7) +#define RT5640_PWR_HP_L_BIT 7 +#define RT5640_PWR_HP_R (0x1 << 6) +#define RT5640_PWR_HP_R_BIT 6 +#define RT5640_PWR_HA (0x1 << 5) +#define RT5640_PWR_HA_BIT 5 +#define RT5640_PWR_VREF2 (0x1 << 4) +#define RT5640_PWR_VREF2_BIT 4 +#define RT5640_PWR_FV2 (0x1 << 3) +#define RT5640_PWR_FV2_BIT 3 +#define RT5640_PWR_LDO2 (0x1 << 2) +#define RT5640_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5640_PWR_BST1 (0x1 << 15) +#define RT5640_PWR_BST1_BIT 15 +#define RT5640_PWR_BST2 (0x1 << 14) +#define RT5640_PWR_BST2_BIT 14 +#define RT5640_PWR_BST3 (0x1 << 13) +#define RT5640_PWR_BST3_BIT 13 +#define RT5640_PWR_BST4 (0x1 << 12) +#define RT5640_PWR_BST4_BIT 12 +#define RT5640_PWR_MB1 (0x1 << 11) +#define RT5640_PWR_MB1_BIT 11 +#define RT5640_PWR_PLL (0x1 << 9) +#define RT5640_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5640_PWR_OM_L (0x1 << 15) +#define RT5640_PWR_OM_L_BIT 15 +#define RT5640_PWR_OM_R (0x1 << 14) +#define RT5640_PWR_OM_R_BIT 14 +#define RT5640_PWR_SM_L (0x1 << 13) +#define RT5640_PWR_SM_L_BIT 13 +#define RT5640_PWR_SM_R (0x1 << 12) +#define RT5640_PWR_SM_R_BIT 12 +#define RT5640_PWR_RM_L (0x1 << 11) +#define RT5640_PWR_RM_L_BIT 11 +#define RT5640_PWR_RM_R (0x1 << 10) +#define RT5640_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5640_PWR_SV_L (0x1 << 15) +#define RT5640_PWR_SV_L_BIT 15 +#define RT5640_PWR_SV_R (0x1 << 14) +#define RT5640_PWR_SV_R_BIT 14 +#define RT5640_PWR_OV_L (0x1 << 13) +#define RT5640_PWR_OV_L_BIT 13 +#define RT5640_PWR_OV_R (0x1 << 12) +#define RT5640_PWR_OV_R_BIT 12 +#define RT5640_PWR_HV_L (0x1 << 11) +#define RT5640_PWR_HV_L_BIT 11 +#define RT5640_PWR_HV_R (0x1 << 10) +#define RT5640_PWR_HV_R_BIT 10 +#define RT5640_PWR_IN_L (0x1 << 9) +#define RT5640_PWR_IN_L_BIT 9 +#define RT5640_PWR_IN_R (0x1 << 8) +#define RT5640_PWR_IN_R_BIT 8 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5640_I2S_MS_MASK (0x1 << 15) +#define RT5640_I2S_MS_SFT 15 +#define RT5640_I2S_MS_M (0x0 << 15) +#define RT5640_I2S_MS_S (0x1 << 15) +#define RT5640_I2S_IF_MASK (0x7 << 12) +#define RT5640_I2S_IF_SFT 12 +#define RT5640_I2S_O_CP_MASK (0x3 << 10) +#define RT5640_I2S_O_CP_SFT 10 +#define RT5640_I2S_O_CP_OFF (0x0 << 10) +#define RT5640_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5640_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5640_I2S_I_CP_MASK (0x3 << 8) +#define RT5640_I2S_I_CP_SFT 8 +#define RT5640_I2S_I_CP_OFF (0x0 << 8) +#define RT5640_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5640_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5640_I2S_BP_MASK (0x1 << 7) +#define RT5640_I2S_BP_SFT 7 +#define RT5640_I2S_BP_NOR (0x0 << 7) +#define RT5640_I2S_BP_INV (0x1 << 7) +#define RT5640_I2S_DL_MASK (0x3 << 2) +#define RT5640_I2S_DL_SFT 2 +#define RT5640_I2S_DL_16 (0x0 << 2) +#define RT5640_I2S_DL_20 (0x1 << 2) +#define RT5640_I2S_DL_24 (0x2 << 2) +#define RT5640_I2S_DL_8 (0x3 << 2) +#define RT5640_I2S_DF_MASK (0x3) +#define RT5640_I2S_DF_SFT 0 +#define RT5640_I2S_DF_I2S (0x0) +#define RT5640_I2S_DF_LEFT (0x1) +#define RT5640_I2S_DF_PCM_A (0x2) +#define RT5640_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5640_I2S2_SDI_MASK (0x1 << 6) +#define RT5640_I2S2_SDI_SFT 6 +#define RT5640_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5640_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5640_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5640_I2S_BCLK_MS1_SFT 15 +#define RT5640_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5640_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5640_I2S_PD1_MASK (0x7 << 12) +#define RT5640_I2S_PD1_SFT 12 +#define RT5640_I2S_PD1_1 (0x0 << 12) +#define RT5640_I2S_PD1_2 (0x1 << 12) +#define RT5640_I2S_PD1_3 (0x2 << 12) +#define RT5640_I2S_PD1_4 (0x3 << 12) +#define RT5640_I2S_PD1_6 (0x4 << 12) +#define RT5640_I2S_PD1_8 (0x5 << 12) +#define RT5640_I2S_PD1_12 (0x6 << 12) +#define RT5640_I2S_PD1_16 (0x7 << 12) +#define RT5640_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5640_I2S_BCLK_MS2_SFT 11 +#define RT5640_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5640_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5640_I2S_PD2_MASK (0x7 << 8) +#define RT5640_I2S_PD2_SFT 8 +#define RT5640_I2S_PD2_1 (0x0 << 8) +#define RT5640_I2S_PD2_2 (0x1 << 8) +#define RT5640_I2S_PD2_3 (0x2 << 8) +#define RT5640_I2S_PD2_4 (0x3 << 8) +#define RT5640_I2S_PD2_6 (0x4 << 8) +#define RT5640_I2S_PD2_8 (0x5 << 8) +#define RT5640_I2S_PD2_12 (0x6 << 8) +#define RT5640_I2S_PD2_16 (0x7 << 8) +#define RT5640_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5640_I2S_BCLK_MS3_SFT 7 +#define RT5640_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5640_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5640_I2S_PD3_MASK (0x7 << 4) +#define RT5640_I2S_PD3_SFT 4 +#define RT5640_I2S_PD3_1 (0x0 << 4) +#define RT5640_I2S_PD3_2 (0x1 << 4) +#define RT5640_I2S_PD3_3 (0x2 << 4) +#define RT5640_I2S_PD3_4 (0x3 << 4) +#define RT5640_I2S_PD3_6 (0x4 << 4) +#define RT5640_I2S_PD3_8 (0x5 << 4) +#define RT5640_I2S_PD3_12 (0x6 << 4) +#define RT5640_I2S_PD3_16 (0x7 << 4) +#define RT5640_DAC_OSR_MASK (0x3 << 2) +#define RT5640_DAC_OSR_SFT 2 +#define RT5640_DAC_OSR_128 (0x0 << 2) +#define RT5640_DAC_OSR_64 (0x1 << 2) +#define RT5640_DAC_OSR_32 (0x2 << 2) +#define RT5640_DAC_OSR_16 (0x3 << 2) +#define RT5640_ADC_OSR_MASK (0x3) +#define RT5640_ADC_OSR_SFT 0 +#define RT5640_ADC_OSR_128 (0x0) +#define RT5640_ADC_OSR_64 (0x1) +#define RT5640_ADC_OSR_32 (0x2) +#define RT5640_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5640_DAC_L_OSR_MASK (0x3 << 14) +#define RT5640_DAC_L_OSR_SFT 14 +#define RT5640_DAC_L_OSR_128 (0x0 << 14) +#define RT5640_DAC_L_OSR_64 (0x1 << 14) +#define RT5640_DAC_L_OSR_32 (0x2 << 14) +#define RT5640_DAC_L_OSR_16 (0x3 << 14) +#define RT5640_ADC_R_OSR_MASK (0x3 << 12) +#define RT5640_ADC_R_OSR_SFT 12 +#define RT5640_ADC_R_OSR_128 (0x0 << 12) +#define RT5640_ADC_R_OSR_64 (0x1 << 12) +#define RT5640_ADC_R_OSR_32 (0x2 << 12) +#define RT5640_ADC_R_OSR_16 (0x3 << 12) +#define RT5640_DAHPF_EN (0x1 << 11) +#define RT5640_DAHPF_EN_SFT 11 +#define RT5640_ADHPF_EN (0x1 << 10) +#define RT5640_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5640_DMIC_1_EN_MASK (0x1 << 15) +#define RT5640_DMIC_1_EN_SFT 15 +#define RT5640_DMIC_1_DIS (0x0 << 15) +#define RT5640_DMIC_1_EN (0x1 << 15) +#define RT5640_DMIC_2_EN_MASK (0x1 << 14) +#define RT5640_DMIC_2_EN_SFT 14 +#define RT5640_DMIC_2_DIS (0x0 << 14) +#define RT5640_DMIC_2_EN (0x1 << 14) +#define RT5640_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5640_DMIC_1L_LH_SFT 13 +#define RT5640_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5640_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5640_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5640_DMIC_1R_LH_SFT 12 +#define RT5640_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5640_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5640_DMIC_1_DP_MASK (0x1 << 11) +#define RT5640_DMIC_1_DP_SFT 11 +#define RT5640_DMIC_1_DP_GPIO3 (0x0 << 11) +#define RT5640_DMIC_1_DP_IN1P (0x1 << 11) +#define RT5640_DMIC_2_DP_MASK (0x1 << 10) +#define RT5640_DMIC_2_DP_SFT 10 +#define RT5640_DMIC_2_DP_GPIO4 (0x0 << 10) +#define RT5640_DMIC_2_DP_IN1N (0x1 << 10) +#define RT5640_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5640_DMIC_2L_LH_SFT 9 +#define RT5640_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5640_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5640_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5640_DMIC_2R_LH_SFT 8 +#define RT5640_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5640_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5640_DMIC_CLK_MASK (0x7 << 5) +#define RT5640_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5640_SCLK_SRC_MASK (0x3 << 14) +#define RT5640_SCLK_SRC_SFT 14 +#define RT5640_SCLK_SRC_MCLK (0x0 << 14) +#define RT5640_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5640_SCLK_SRC_PLL1T (0x2 << 14) +#define RT5640_SCLK_SRC_RCCLK (0x3 << 14) /* 15MHz */ +#define RT5640_PLL1_SRC_MASK (0x3 << 12) +#define RT5640_PLL1_SRC_SFT 12 +#define RT5640_PLL1_SRC_MCLK (0x0 << 12) +#define RT5640_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5640_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5640_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5640_PLL1_PD_MASK (0x1 << 3) +#define RT5640_PLL1_PD_SFT 3 +#define RT5640_PLL1_PD_1 (0x0 << 3) +#define RT5640_PLL1_PD_2 (0x1 << 3) + +#define RT5640_PLL_INP_MAX 40000000 +#define RT5640_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5640_PLL_N_MAX 0x1ff +#define RT5640_PLL_N_MASK (RT5640_PLL_N_MAX << 7) +#define RT5640_PLL_N_SFT 7 +#define RT5640_PLL_K_MAX 0x1f +#define RT5640_PLL_K_MASK (RT5640_PLL_K_MAX) +#define RT5640_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5640_PLL_M_MAX 0xf +#define RT5640_PLL_M_MASK (RT5640_PLL_M_MAX << 12) +#define RT5640_PLL_M_SFT 12 +#define RT5640_PLL_M_BP (0x1 << 11) +#define RT5640_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5640_STO_T_MASK (0x1 << 15) +#define RT5640_STO_T_SFT 15 +#define RT5640_STO_T_SCLK (0x0 << 15) +#define RT5640_STO_T_LRCK1 (0x1 << 15) +#define RT5640_M1_T_MASK (0x1 << 14) +#define RT5640_M1_T_SFT 14 +#define RT5640_M1_T_I2S2 (0x0 << 14) +#define RT5640_M1_T_I2S2_D3 (0x1 << 14) +#define RT5640_I2S2_F_MASK (0x1 << 12) +#define RT5640_I2S2_F_SFT 12 +#define RT5640_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5640_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5640_DMIC_1_M_MASK (0x1 << 9) +#define RT5640_DMIC_1_M_SFT 9 +#define RT5640_DMIC_1_M_NOR (0x0 << 9) +#define RT5640_DMIC_1_M_ASYN (0x1 << 9) +#define RT5640_DMIC_2_M_MASK (0x1 << 8) +#define RT5640_DMIC_2_M_SFT 8 +#define RT5640_DMIC_2_M_NOR (0x0 << 8) +#define RT5640_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5640_MDA_L_M_MASK (0x1 << 15) +#define RT5640_MDA_L_M_SFT 15 +#define RT5640_MDA_L_M_NOR (0x0 << 15) +#define RT5640_MDA_L_M_ASYN (0x1 << 15) +#define RT5640_MDA_R_M_MASK (0x1 << 14) +#define RT5640_MDA_R_M_SFT 14 +#define RT5640_MDA_R_M_NOR (0x0 << 14) +#define RT5640_MDA_R_M_ASYN (0x1 << 14) +#define RT5640_MAD_L_M_MASK (0x1 << 13) +#define RT5640_MAD_L_M_SFT 13 +#define RT5640_MAD_L_M_NOR (0x0 << 13) +#define RT5640_MAD_L_M_ASYN (0x1 << 13) +#define RT5640_MAD_R_M_MASK (0x1 << 12) +#define RT5640_MAD_R_M_SFT 12 +#define RT5640_MAD_R_M_NOR (0x0 << 12) +#define RT5640_MAD_R_M_ASYN (0x1 << 12) +#define RT5640_ADC_M_MASK (0x1 << 11) +#define RT5640_ADC_M_SFT 11 +#define RT5640_ADC_M_NOR (0x0 << 11) +#define RT5640_ADC_M_ASYN (0x1 << 11) +#define RT5640_STO_DAC_M_MASK (0x1 << 5) +#define RT5640_STO_DAC_M_SFT 5 +#define RT5640_STO_DAC_M_NOR (0x0 << 5) +#define RT5640_STO_DAC_M_ASYN (0x1 << 5) +#define RT5640_I2S1_R_D_MASK (0x1 << 4) +#define RT5640_I2S1_R_D_SFT 4 +#define RT5640_I2S1_R_D_DIS (0x0 << 4) +#define RT5640_I2S1_R_D_EN (0x1 << 4) +#define RT5640_I2S2_R_D_MASK (0x1 << 3) +#define RT5640_I2S2_R_D_SFT 3 +#define RT5640_I2S2_R_D_DIS (0x0 << 3) +#define RT5640_I2S2_R_D_EN (0x1 << 3) +#define RT5640_PRE_SCLK_MASK (0x3) +#define RT5640_PRE_SCLK_SFT 0 +#define RT5640_PRE_SCLK_512 (0x0) +#define RT5640_PRE_SCLK_1024 (0x1) +#define RT5640_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5640_I2S1_RATE_MASK (0xf << 12) +#define RT5640_I2S1_RATE_SFT 12 +#define RT5640_I2S2_RATE_MASK (0xf << 8) +#define RT5640_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5640_I2S1_PD_MASK (0x7 << 12) +#define RT5640_I2S1_PD_SFT 12 +#define RT5640_I2S2_PD_MASK (0x7 << 8) +#define RT5640_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5640_HP_OVCD_MASK (0x1 << 10) +#define RT5640_HP_OVCD_SFT 10 +#define RT5640_HP_OVCD_DIS (0x0 << 10) +#define RT5640_HP_OVCD_EN (0x1 << 10) +#define RT5640_HP_OC_TH_MASK (0x3 << 8) +#define RT5640_HP_OC_TH_SFT 8 +#define RT5640_HP_OC_TH_90 (0x0 << 8) +#define RT5640_HP_OC_TH_105 (0x1 << 8) +#define RT5640_HP_OC_TH_120 (0x2 << 8) +#define RT5640_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5640_CLSD_OC_MASK (0x1 << 9) +#define RT5640_CLSD_OC_SFT 9 +#define RT5640_CLSD_OC_PU (0x0 << 9) +#define RT5640_CLSD_OC_PD (0x1 << 9) +#define RT5640_AUTO_PD_MASK (0x1 << 8) +#define RT5640_AUTO_PD_SFT 8 +#define RT5640_AUTO_PD_DIS (0x0 << 8) +#define RT5640_AUTO_PD_EN (0x1 << 8) +#define RT5640_CLSD_OC_TH_MASK (0x3f) +#define RT5640_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5640_CLSD_RATIO_MASK (0xf << 12) +#define RT5640_CLSD_RATIO_SFT 12 +#define RT5640_CLSD_OM_MASK (0x1 << 11) +#define RT5640_CLSD_OM_SFT 11 +#define RT5640_CLSD_OM_MONO (0x0 << 11) +#define RT5640_CLSD_OM_STO (0x1 << 11) +#define RT5640_CLSD_SCH_MASK (0x1 << 10) +#define RT5640_CLSD_SCH_SFT 10 +#define RT5640_CLSD_SCH_L (0x0 << 10) +#define RT5640_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5640_SMT_TRIG_MASK (0x1 << 15) +#define RT5640_SMT_TRIG_SFT 15 +#define RT5640_SMT_TRIG_DIS (0x0 << 15) +#define RT5640_SMT_TRIG_EN (0x1 << 15) +#define RT5640_HP_L_SMT_MASK (0x1 << 9) +#define RT5640_HP_L_SMT_SFT 9 +#define RT5640_HP_L_SMT_DIS (0x0 << 9) +#define RT5640_HP_L_SMT_EN (0x1 << 9) +#define RT5640_HP_R_SMT_MASK (0x1 << 8) +#define RT5640_HP_R_SMT_SFT 8 +#define RT5640_HP_R_SMT_DIS (0x0 << 8) +#define RT5640_HP_R_SMT_EN (0x1 << 8) +#define RT5640_HP_CD_PD_MASK (0x1 << 7) +#define RT5640_HP_CD_PD_SFT 7 +#define RT5640_HP_CD_PD_DIS (0x0 << 7) +#define RT5640_HP_CD_PD_EN (0x1 << 7) +#define RT5640_RSTN_MASK (0x1 << 6) +#define RT5640_RSTN_SFT 6 +#define RT5640_RSTN_DIS (0x0 << 6) +#define RT5640_RSTN_EN (0x1 << 6) +#define RT5640_RSTP_MASK (0x1 << 5) +#define RT5640_RSTP_SFT 5 +#define RT5640_RSTP_DIS (0x0 << 5) +#define RT5640_RSTP_EN (0x1 << 5) +#define RT5640_HP_CO_MASK (0x1 << 4) +#define RT5640_HP_CO_SFT 4 +#define RT5640_HP_CO_DIS (0x0 << 4) +#define RT5640_HP_CO_EN (0x1 << 4) +#define RT5640_HP_CP_MASK (0x1 << 3) +#define RT5640_HP_CP_SFT 3 +#define RT5640_HP_CP_PD (0x0 << 3) +#define RT5640_HP_CP_PU (0x1 << 3) +#define RT5640_HP_SG_MASK (0x1 << 2) +#define RT5640_HP_SG_SFT 2 +#define RT5640_HP_SG_DIS (0x0 << 2) +#define RT5640_HP_SG_EN (0x1 << 2) +#define RT5640_HP_DP_MASK (0x1 << 1) +#define RT5640_HP_DP_SFT 1 +#define RT5640_HP_DP_PD (0x0 << 1) +#define RT5640_HP_DP_PU (0x1 << 1) +#define RT5640_HP_CB_MASK (0x1) +#define RT5640_HP_CB_SFT 0 +#define RT5640_HP_CB_PD (0x0) +#define RT5640_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5640_DEPOP_MASK (0x1 << 13) +#define RT5640_DEPOP_SFT 13 +#define RT5640_DEPOP_AUTO (0x0 << 13) +#define RT5640_DEPOP_MAN (0x1 << 13) +#define RT5640_RAMP_MASK (0x1 << 12) +#define RT5640_RAMP_SFT 12 +#define RT5640_RAMP_DIS (0x0 << 12) +#define RT5640_RAMP_EN (0x1 << 12) +#define RT5640_BPS_MASK (0x1 << 11) +#define RT5640_BPS_SFT 11 +#define RT5640_BPS_DIS (0x0 << 11) +#define RT5640_BPS_EN (0x1 << 11) +#define RT5640_FAST_UPDN_MASK (0x1 << 10) +#define RT5640_FAST_UPDN_SFT 10 +#define RT5640_FAST_UPDN_DIS (0x0 << 10) +#define RT5640_FAST_UPDN_EN (0x1 << 10) +#define RT5640_MRES_MASK (0x3 << 8) +#define RT5640_MRES_SFT 8 +#define RT5640_MRES_15MO (0x0 << 8) +#define RT5640_MRES_25MO (0x1 << 8) +#define RT5640_MRES_35MO (0x2 << 8) +#define RT5640_MRES_45MO (0x3 << 8) +#define RT5640_VLO_MASK (0x1 << 7) +#define RT5640_VLO_SFT 7 +#define RT5640_VLO_3V (0x0 << 7) +#define RT5640_VLO_32V (0x1 << 7) +#define RT5640_DIG_DP_MASK (0x1 << 6) +#define RT5640_DIG_DP_SFT 6 +#define RT5640_DIG_DP_DIS (0x0 << 6) +#define RT5640_DIG_DP_EN (0x1 << 6) +#define RT5640_DP_TH_MASK (0x3 << 4) +#define RT5640_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5640_CP_SYS_MASK (0x7 << 12) +#define RT5640_CP_SYS_SFT 12 +#define RT5640_CP_FQ1_MASK (0x7 << 8) +#define RT5640_CP_FQ1_SFT 8 +#define RT5640_CP_FQ2_MASK (0x7 << 4) +#define RT5640_CP_FQ2_SFT 4 +#define RT5640_CP_FQ3_MASK (0x7) +#define RT5640_CP_FQ3_SFT 0 + +/* HPOUT charge pump (0x91) */ +#define RT5640_OSW_L_MASK (0x1 << 11) +#define RT5640_OSW_L_SFT 11 +#define RT5640_OSW_L_DIS (0x0 << 11) +#define RT5640_OSW_L_EN (0x1 << 11) +#define RT5640_OSW_R_MASK (0x1 << 10) +#define RT5640_OSW_R_SFT 10 +#define RT5640_OSW_R_DIS (0x0 << 10) +#define RT5640_OSW_R_EN (0x1 << 10) +#define RT5640_PM_HP_MASK (0x3 << 8) +#define RT5640_PM_HP_SFT 8 +#define RT5640_PM_HP_LV (0x0 << 8) +#define RT5640_PM_HP_MV (0x1 << 8) +#define RT5640_PM_HP_HV (0x2 << 8) +#define RT5640_IB_HP_MASK (0x3 << 6) +#define RT5640_IB_HP_SFT 6 +#define RT5640_IB_HP_125IL (0x0 << 6) +#define RT5640_IB_HP_25IL (0x1 << 6) +#define RT5640_IB_HP_5IL (0x2 << 6) +#define RT5640_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5640_PVDD_DET_MASK (0x1 << 15) +#define RT5640_PVDD_DET_SFT 15 +#define RT5640_PVDD_DET_DIS (0x0 << 15) +#define RT5640_PVDD_DET_EN (0x1 << 15) +#define RT5640_SPK_AG_MASK (0x1 << 14) +#define RT5640_SPK_AG_SFT 14 +#define RT5640_SPK_AG_DIS (0x0 << 14) +#define RT5640_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5640_MIC1_BS_MASK (0x1 << 15) +#define RT5640_MIC1_BS_SFT 15 +#define RT5640_MIC1_BS_9AV (0x0 << 15) +#define RT5640_MIC1_BS_75AV (0x1 << 15) +#define RT5640_MIC2_BS_MASK (0x1 << 14) +#define RT5640_MIC2_BS_SFT 14 +#define RT5640_MIC2_BS_9AV (0x0 << 14) +#define RT5640_MIC2_BS_75AV (0x1 << 14) +#define RT5640_MIC1_CLK_MASK (0x1 << 13) +#define RT5640_MIC1_CLK_SFT 13 +#define RT5640_MIC1_CLK_DIS (0x0 << 13) +#define RT5640_MIC1_CLK_EN (0x1 << 13) +#define RT5640_MIC2_CLK_MASK (0x1 << 12) +#define RT5640_MIC2_CLK_SFT 12 +#define RT5640_MIC2_CLK_DIS (0x0 << 12) +#define RT5640_MIC2_CLK_EN (0x1 << 12) +#define RT5640_MIC1_OVCD_MASK (0x1 << 11) +#define RT5640_MIC1_OVCD_SFT 11 +#define RT5640_MIC1_OVCD_DIS (0x0 << 11) +#define RT5640_MIC1_OVCD_EN (0x1 << 11) +#define RT5640_MIC1_OVTH_MASK (0x3 << 9) +#define RT5640_MIC1_OVTH_SFT 9 +#define RT5640_MIC1_OVTH_600UA (0x0 << 9) +#define RT5640_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5640_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5640_MIC2_OVCD_MASK (0x1 << 8) +#define RT5640_MIC2_OVCD_SFT 8 +#define RT5640_MIC2_OVCD_DIS (0x0 << 8) +#define RT5640_MIC2_OVCD_EN (0x1 << 8) +#define RT5640_MIC2_OVTH_MASK (0x3 << 6) +#define RT5640_MIC2_OVTH_SFT 6 +#define RT5640_MIC2_OVTH_600UA (0x0 << 6) +#define RT5640_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5640_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5640_PWR_MB_MASK (0x1 << 5) +#define RT5640_PWR_MB_SFT 5 +#define RT5640_PWR_MB_PD (0x0 << 5) +#define RT5640_PWR_MB_PU (0x1 << 5) +#define RT5640_PWR_CLK25M_MASK (0x1 << 4) +#define RT5640_PWR_CLK25M_SFT 4 +#define RT5640_PWR_CLK25M_PD (0x0 << 4) +#define RT5640_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5640_EQ_SRC_MASK (0x1 << 15) +#define RT5640_EQ_SRC_SFT 15 +#define RT5640_EQ_SRC_DAC (0x0 << 15) +#define RT5640_EQ_SRC_ADC (0x1 << 15) +#define RT5640_EQ_UPD (0x1 << 14) +#define RT5640_EQ_UPD_BIT 14 +#define RT5640_EQ_CD_MASK (0x1 << 13) +#define RT5640_EQ_CD_SFT 13 +#define RT5640_EQ_CD_DIS (0x0 << 13) +#define RT5640_EQ_CD_EN (0x1 << 13) +#define RT5640_EQ_DITH_MASK (0x3 << 8) +#define RT5640_EQ_DITH_SFT 8 +#define RT5640_EQ_DITH_NOR (0x0 << 8) +#define RT5640_EQ_DITH_LSB (0x1 << 8) +#define RT5640_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5640_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5640_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5640_EQ_HPF1_M_SFT 8 +#define RT5640_EQ_HPF1_M_HI (0x0 << 8) +#define RT5640_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5640_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5640_EQ_LPF1_M_SFT 7 +#define RT5640_EQ_LPF1_M_LO (0x0 << 7) +#define RT5640_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5640_EQ_HPF2_MASK (0x1 << 6) +#define RT5640_EQ_HPF2_SFT 6 +#define RT5640_EQ_HPF2_DIS (0x0 << 6) +#define RT5640_EQ_HPF2_EN (0x1 << 6) +#define RT5640_EQ_HPF1_MASK (0x1 << 5) +#define RT5640_EQ_HPF1_SFT 5 +#define RT5640_EQ_HPF1_DIS (0x0 << 5) +#define RT5640_EQ_HPF1_EN (0x1 << 5) +#define RT5640_EQ_BPF4_MASK (0x1 << 4) +#define RT5640_EQ_BPF4_SFT 4 +#define RT5640_EQ_BPF4_DIS (0x0 << 4) +#define RT5640_EQ_BPF4_EN (0x1 << 4) +#define RT5640_EQ_BPF3_MASK (0x1 << 3) +#define RT5640_EQ_BPF3_SFT 3 +#define RT5640_EQ_BPF3_DIS (0x0 << 3) +#define RT5640_EQ_BPF3_EN (0x1 << 3) +#define RT5640_EQ_BPF2_MASK (0x1 << 2) +#define RT5640_EQ_BPF2_SFT 2 +#define RT5640_EQ_BPF2_DIS (0x0 << 2) +#define RT5640_EQ_BPF2_EN (0x1 << 2) +#define RT5640_EQ_BPF1_MASK (0x1 << 1) +#define RT5640_EQ_BPF1_SFT 1 +#define RT5640_EQ_BPF1_DIS (0x0 << 1) +#define RT5640_EQ_BPF1_EN (0x1 << 1) +#define RT5640_EQ_LPF_MASK (0x1) +#define RT5640_EQ_LPF_SFT 0 +#define RT5640_EQ_LPF_DIS (0x0) +#define RT5640_EQ_LPF_EN (0x1) + +/* Memory Test (0xb2) */ +#define RT5640_MT_MASK (0x1 << 15) +#define RT5640_MT_SFT 15 +#define RT5640_MT_DIS (0x0 << 15) +#define RT5640_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5640_DRC_AGC_P_MASK (0x1 << 15) +#define RT5640_DRC_AGC_P_SFT 15 +#define RT5640_DRC_AGC_P_DAC (0x0 << 15) +#define RT5640_DRC_AGC_P_ADC (0x1 << 15) +#define RT5640_DRC_AGC_MASK (0x1 << 14) +#define RT5640_DRC_AGC_SFT 14 +#define RT5640_DRC_AGC_DIS (0x0 << 14) +#define RT5640_DRC_AGC_EN (0x1 << 14) +#define RT5640_DRC_AGC_UPD (0x1 << 13) +#define RT5640_DRC_AGC_UPD_BIT 13 +#define RT5640_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5640_DRC_AGC_AR_SFT 8 +#define RT5640_DRC_AGC_R_MASK (0x7 << 5) +#define RT5640_DRC_AGC_R_SFT 5 +#define RT5640_DRC_AGC_R_48K (0x1 << 5) +#define RT5640_DRC_AGC_R_96K (0x2 << 5) +#define RT5640_DRC_AGC_R_192K (0x3 << 5) +#define RT5640_DRC_AGC_R_441K (0x5 << 5) +#define RT5640_DRC_AGC_R_882K (0x6 << 5) +#define RT5640_DRC_AGC_R_1764K (0x7 << 5) +#define RT5640_DRC_AGC_RC_MASK (0x1f) +#define RT5640_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5640_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5640_DRC_AGC_POB_SFT 8 +#define RT5640_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5640_DRC_AGC_CP_SFT 7 +#define RT5640_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5640_DRC_AGC_CP_EN (0x1 << 7) +#define RT5640_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5640_DRC_AGC_CPR_SFT 5 +#define RT5640_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5640_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5640_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5640_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5640_DRC_AGC_PRB_MASK (0x1f) +#define RT5640_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5640_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5640_DRC_AGC_NGB_SFT 12 +#define RT5640_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5640_DRC_AGC_TAR_SFT 7 +#define RT5640_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5640_DRC_AGC_NG_SFT 6 +#define RT5640_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5640_DRC_AGC_NG_EN (0x1 << 6) +#define RT5640_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5640_DRC_AGC_NGH_SFT 5 +#define RT5640_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5640_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5640_DRC_AGC_NGT_MASK (0x1f) +#define RT5640_DRC_AGC_NGT_SFT 0 + +/* ANC Control 1 (0xb8) */ +#define RT5640_ANC_M_MASK (0x1 << 15) +#define RT5640_ANC_M_SFT 15 +#define RT5640_ANC_M_NOR (0x0 << 15) +#define RT5640_ANC_M_REV (0x1 << 15) +#define RT5640_ANC_MASK (0x1 << 14) +#define RT5640_ANC_SFT 14 +#define RT5640_ANC_DIS (0x0 << 14) +#define RT5640_ANC_EN (0x1 << 14) +#define RT5640_ANC_MD_MASK (0x3 << 12) +#define RT5640_ANC_MD_SFT 12 +#define RT5640_ANC_MD_DIS (0x0 << 12) +#define RT5640_ANC_MD_67MS (0x1 << 12) +#define RT5640_ANC_MD_267MS (0x2 << 12) +#define RT5640_ANC_MD_1067MS (0x3 << 12) +#define RT5640_ANC_SN_MASK (0x1 << 11) +#define RT5640_ANC_SN_SFT 11 +#define RT5640_ANC_SN_DIS (0x0 << 11) +#define RT5640_ANC_SN_EN (0x1 << 11) +#define RT5640_ANC_CLK_MASK (0x1 << 10) +#define RT5640_ANC_CLK_SFT 10 +#define RT5640_ANC_CLK_ANC (0x0 << 10) +#define RT5640_ANC_CLK_REG (0x1 << 10) +#define RT5640_ANC_ZCD_MASK (0x3 << 8) +#define RT5640_ANC_ZCD_SFT 8 +#define RT5640_ANC_ZCD_DIS (0x0 << 8) +#define RT5640_ANC_ZCD_T1 (0x1 << 8) +#define RT5640_ANC_ZCD_T2 (0x2 << 8) +#define RT5640_ANC_ZCD_WT (0x3 << 8) +#define RT5640_ANC_CS_MASK (0x1 << 7) +#define RT5640_ANC_CS_SFT 7 +#define RT5640_ANC_CS_DIS (0x0 << 7) +#define RT5640_ANC_CS_EN (0x1 << 7) +#define RT5640_ANC_SW_MASK (0x1 << 6) +#define RT5640_ANC_SW_SFT 6 +#define RT5640_ANC_SW_NOR (0x0 << 6) +#define RT5640_ANC_SW_AUTO (0x1 << 6) +#define RT5640_ANC_CO_L_MASK (0x3f) +#define RT5640_ANC_CO_L_SFT 0 + +/* ANC Control 2 (0xb6) */ +#define RT5640_ANC_FG_R_MASK (0xf << 12) +#define RT5640_ANC_FG_R_SFT 12 +#define RT5640_ANC_FG_L_MASK (0xf << 8) +#define RT5640_ANC_FG_L_SFT 8 +#define RT5640_ANC_CG_R_MASK (0xf << 4) +#define RT5640_ANC_CG_R_SFT 4 +#define RT5640_ANC_CG_L_MASK (0xf) +#define RT5640_ANC_CG_L_SFT 0 + +/* ANC Control 3 (0xb6) */ +#define RT5640_ANC_CD_MASK (0x1 << 6) +#define RT5640_ANC_CD_SFT 6 +#define RT5640_ANC_CD_BOTH (0x0 << 6) +#define RT5640_ANC_CD_IND (0x1 << 6) +#define RT5640_ANC_CO_R_MASK (0x3f) +#define RT5640_ANC_CO_R_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5640_JD_MASK (0x7 << 13) +#define RT5640_JD_SFT 13 +#define RT5640_JD_DIS (0x0 << 13) +#define RT5640_JD_GPIO1 (0x1 << 13) +#define RT5640_JD_JD1_IN4P (0x2 << 13) +#define RT5640_JD_JD2_IN4N (0x3 << 13) +#define RT5640_JD_GPIO2 (0x4 << 13) +#define RT5640_JD_GPIO3 (0x5 << 13) +#define RT5640_JD_GPIO4 (0x6 << 13) +#define RT5640_JD_HP_MASK (0x1 << 11) +#define RT5640_JD_HP_SFT 11 +#define RT5640_JD_HP_DIS (0x0 << 11) +#define RT5640_JD_HP_EN (0x1 << 11) +#define RT5640_JD_HP_TRG_MASK (0x1 << 10) +#define RT5640_JD_HP_TRG_SFT 10 +#define RT5640_JD_HP_TRG_LO (0x0 << 10) +#define RT5640_JD_HP_TRG_HI (0x1 << 10) +#define RT5640_JD_SPL_MASK (0x1 << 9) +#define RT5640_JD_SPL_SFT 9 +#define RT5640_JD_SPL_DIS (0x0 << 9) +#define RT5640_JD_SPL_EN (0x1 << 9) +#define RT5640_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5640_JD_SPL_TRG_SFT 8 +#define RT5640_JD_SPL_TRG_LO (0x0 << 8) +#define RT5640_JD_SPL_TRG_HI (0x1 << 8) +#define RT5640_JD_SPR_MASK (0x1 << 7) +#define RT5640_JD_SPR_SFT 7 +#define RT5640_JD_SPR_DIS (0x0 << 7) +#define RT5640_JD_SPR_EN (0x1 << 7) +#define RT5640_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5640_JD_SPR_TRG_SFT 6 +#define RT5640_JD_SPR_TRG_LO (0x0 << 6) +#define RT5640_JD_SPR_TRG_HI (0x1 << 6) +#define RT5640_JD_MO_MASK (0x1 << 5) +#define RT5640_JD_MO_SFT 5 +#define RT5640_JD_MO_DIS (0x0 << 5) +#define RT5640_JD_MO_EN (0x1 << 5) +#define RT5640_JD_MO_TRG_MASK (0x1 << 4) +#define RT5640_JD_MO_TRG_SFT 4 +#define RT5640_JD_MO_TRG_LO (0x0 << 4) +#define RT5640_JD_MO_TRG_HI (0x1 << 4) +#define RT5640_JD_LO_MASK (0x1 << 3) +#define RT5640_JD_LO_SFT 3 +#define RT5640_JD_LO_DIS (0x0 << 3) +#define RT5640_JD_LO_EN (0x1 << 3) +#define RT5640_JD_LO_TRG_MASK (0x1 << 2) +#define RT5640_JD_LO_TRG_SFT 2 +#define RT5640_JD_LO_TRG_LO (0x0 << 2) +#define RT5640_JD_LO_TRG_HI (0x1 << 2) +#define RT5640_JD1_IN4P_MASK (0x1 << 1) +#define RT5640_JD1_IN4P_SFT 1 +#define RT5640_JD1_IN4P_DIS (0x0 << 1) +#define RT5640_JD1_IN4P_EN (0x1 << 1) +#define RT5640_JD2_IN4N_MASK (0x1) +#define RT5640_JD2_IN4N_SFT 0 +#define RT5640_JD2_IN4N_DIS (0x0) +#define RT5640_JD2_IN4N_EN (0x1) + +/* Jack detect for ANC (0xbc) */ +#define RT5640_ANC_DET_MASK (0x3 << 4) +#define RT5640_ANC_DET_SFT 4 +#define RT5640_ANC_DET_DIS (0x0 << 4) +#define RT5640_ANC_DET_MB1 (0x1 << 4) +#define RT5640_ANC_DET_MB2 (0x2 << 4) +#define RT5640_ANC_DET_JD (0x3 << 4) +#define RT5640_AD_TRG_MASK (0x1 << 3) +#define RT5640_AD_TRG_SFT 3 +#define RT5640_AD_TRG_LO (0x0 << 3) +#define RT5640_AD_TRG_HI (0x1 << 3) +#define RT5640_ANCM_DET_MASK (0x3 << 4) +#define RT5640_ANCM_DET_SFT 4 +#define RT5640_ANCM_DET_DIS (0x0 << 4) +#define RT5640_ANCM_DET_MB1 (0x1 << 4) +#define RT5640_ANCM_DET_MB2 (0x2 << 4) +#define RT5640_ANCM_DET_JD (0x3 << 4) +#define RT5640_AMD_TRG_MASK (0x1 << 3) +#define RT5640_AMD_TRG_SFT 3 +#define RT5640_AMD_TRG_LO (0x0 << 3) +#define RT5640_AMD_TRG_HI (0x1 << 3) + +/* IRQ Control 1 (0xbd) */ +#define RT5640_IRQ_JD_MASK (0x1 << 15) +#define RT5640_IRQ_JD_SFT 15 +#define RT5640_IRQ_JD_BP (0x0 << 15) +#define RT5640_IRQ_JD_NOR (0x1 << 15) +#define RT5640_IRQ_OT_MASK (0x1 << 14) +#define RT5640_IRQ_OT_SFT 14 +#define RT5640_IRQ_OT_BP (0x0 << 14) +#define RT5640_IRQ_OT_NOR (0x1 << 14) +#define RT5640_JD_STKY_MASK (0x1 << 13) +#define RT5640_JD_STKY_SFT 13 +#define RT5640_JD_STKY_DIS (0x0 << 13) +#define RT5640_JD_STKY_EN (0x1 << 13) +#define RT5640_OT_STKY_MASK (0x1 << 12) +#define RT5640_OT_STKY_SFT 12 +#define RT5640_OT_STKY_DIS (0x0 << 12) +#define RT5640_OT_STKY_EN (0x1 << 12) +#define RT5640_JD_P_MASK (0x1 << 11) +#define RT5640_JD_P_SFT 11 +#define RT5640_JD_P_NOR (0x0 << 11) +#define RT5640_JD_P_INV (0x1 << 11) +#define RT5640_OT_P_MASK (0x1 << 10) +#define RT5640_OT_P_SFT 10 +#define RT5640_OT_P_NOR (0x0 << 10) +#define RT5640_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5640_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5640_IRQ_MB1_OC_SFT 15 +#define RT5640_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5640_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5640_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5640_IRQ_MB2_OC_SFT 14 +#define RT5640_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5640_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5640_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5640_MB1_OC_STKY_SFT 11 +#define RT5640_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5640_MB1_OC_STKY_EN (0x1 << 11) +#define RT5640_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5640_MB2_OC_STKY_SFT 10 +#define RT5640_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5640_MB2_OC_STKY_EN (0x1 << 10) +#define RT5640_MB1_OC_P_MASK (0x1 << 7) +#define RT5640_MB1_OC_P_SFT 7 +#define RT5640_MB1_OC_P_NOR (0x0 << 7) +#define RT5640_MB1_OC_P_INV (0x1 << 7) +#define RT5640_MB2_OC_P_MASK (0x1 << 6) +#define RT5640_MB2_OC_P_SFT 6 +#define RT5640_MB2_OC_P_NOR (0x0 << 6) +#define RT5640_MB2_OC_P_INV (0x1 << 6) +#define RT5640_MB1_OC_CLR (0x1 << 3) +#define RT5640_MB1_OC_CLR_SFT 3 +#define RT5640_MB2_OC_CLR (0x1 << 2) +#define RT5640_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5640_GP1_PIN_MASK (0x1 << 15) +#define RT5640_GP1_PIN_SFT 15 +#define RT5640_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5640_GP1_PIN_IRQ (0x1 << 15) +#define RT5640_GP2_PIN_MASK (0x1 << 14) +#define RT5640_GP2_PIN_SFT 14 +#define RT5640_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5640_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5640_GP3_PIN_MASK (0x3 << 12) +#define RT5640_GP3_PIN_SFT 12 +#define RT5640_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5640_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5640_GP3_PIN_IRQ (0x2 << 12) +#define RT5640_GP4_PIN_MASK (0x1 << 11) +#define RT5640_GP4_PIN_SFT 11 +#define RT5640_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5640_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5640_DP_SIG_MASK (0x1 << 10) +#define RT5640_DP_SIG_SFT 10 +#define RT5640_DP_SIG_TEST (0x0 << 10) +#define RT5640_DP_SIG_AP (0x1 << 10) +#define RT5640_GPIO_M_MASK (0x1 << 9) +#define RT5640_GPIO_M_SFT 9 +#define RT5640_GPIO_M_FLT (0x0 << 9) +#define RT5640_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5640_GP4_PF_MASK (0x1 << 11) +#define RT5640_GP4_PF_SFT 11 +#define RT5640_GP4_PF_IN (0x0 << 11) +#define RT5640_GP4_PF_OUT (0x1 << 11) +#define RT5640_GP4_OUT_MASK (0x1 << 10) +#define RT5640_GP4_OUT_SFT 10 +#define RT5640_GP4_OUT_LO (0x0 << 10) +#define RT5640_GP4_OUT_HI (0x1 << 10) +#define RT5640_GP4_P_MASK (0x1 << 9) +#define RT5640_GP4_P_SFT 9 +#define RT5640_GP4_P_NOR (0x0 << 9) +#define RT5640_GP4_P_INV (0x1 << 9) +#define RT5640_GP3_PF_MASK (0x1 << 8) +#define RT5640_GP3_PF_SFT 8 +#define RT5640_GP3_PF_IN (0x0 << 8) +#define RT5640_GP3_PF_OUT (0x1 << 8) +#define RT5640_GP3_OUT_MASK (0x1 << 7) +#define RT5640_GP3_OUT_SFT 7 +#define RT5640_GP3_OUT_LO (0x0 << 7) +#define RT5640_GP3_OUT_HI (0x1 << 7) +#define RT5640_GP3_P_MASK (0x1 << 6) +#define RT5640_GP3_P_SFT 6 +#define RT5640_GP3_P_NOR (0x0 << 6) +#define RT5640_GP3_P_INV (0x1 << 6) +#define RT5640_GP2_PF_MASK (0x1 << 5) +#define RT5640_GP2_PF_SFT 5 +#define RT5640_GP2_PF_IN (0x0 << 5) +#define RT5640_GP2_PF_OUT (0x1 << 5) +#define RT5640_GP2_OUT_MASK (0x1 << 4) +#define RT5640_GP2_OUT_SFT 4 +#define RT5640_GP2_OUT_LO (0x0 << 4) +#define RT5640_GP2_OUT_HI (0x1 << 4) +#define RT5640_GP2_P_MASK (0x1 << 3) +#define RT5640_GP2_P_SFT 3 +#define RT5640_GP2_P_NOR (0x0 << 3) +#define RT5640_GP2_P_INV (0x1 << 3) +#define RT5640_GP1_PF_MASK (0x1 << 2) +#define RT5640_GP1_PF_SFT 2 +#define RT5640_GP1_PF_IN (0x0 << 2) +#define RT5640_GP1_PF_OUT (0x1 << 2) +#define RT5640_GP1_OUT_MASK (0x1 << 1) +#define RT5640_GP1_OUT_SFT 1 +#define RT5640_GP1_OUT_LO (0x0 << 1) +#define RT5640_GP1_OUT_HI (0x1 << 1) +#define RT5640_GP1_P_MASK (0x1) +#define RT5640_GP1_P_SFT 0 +#define RT5640_GP1_P_NOR (0x0) +#define RT5640_GP1_P_INV (0x1) + +/* FM34-500 Register Control 1 (0xc4) */ +#define RT5640_DSP_ADD_SFT 0 + +/* FM34-500 Register Control 2 (0xc5) */ +#define RT5640_DSP_DAT_SFT 0 + +/* FM34-500 Register Control 3 (0xc6) */ +#define RT5640_DSP_BUSY_MASK (0x1 << 15) +#define RT5640_DSP_BUSY_BIT 15 +#define RT5640_DSP_DS_MASK (0x1 << 14) +#define RT5640_DSP_DS_SFT 14 +#define RT5640_DSP_DS_FM3010 (0x1 << 14) +#define RT5640_DSP_DS_TEMP (0x1 << 14) +#define RT5640_DSP_CLK_MASK (0x3 << 12) +#define RT5640_DSP_CLK_SFT 12 +#define RT5640_DSP_CLK_384K (0x0 << 12) +#define RT5640_DSP_CLK_192K (0x1 << 12) +#define RT5640_DSP_CLK_96K (0x2 << 12) +#define RT5640_DSP_CLK_64K (0x3 << 12) +#define RT5640_DSP_PD_PIN_MASK (0x1 << 11) +#define RT5640_DSP_PD_PIN_SFT 11 +#define RT5640_DSP_PD_PIN_LO (0x0 << 11) +#define RT5640_DSP_PD_PIN_HI (0x1 << 11) +#define RT5640_DSP_RST_PIN_MASK (0x1 << 10) +#define RT5640_DSP_RST_PIN_SFT 10 +#define RT5640_DSP_RST_PIN_LO (0x0 << 10) +#define RT5640_DSP_RST_PIN_HI (0x1 << 10) +#define RT5640_DSP_R_EN (0x1 << 9) +#define RT5640_DSP_R_EN_BIT 9 +#define RT5640_DSP_W_EN (0x1 << 8) +#define RT5640_DSP_W_EN_BIT 8 +#define RT5640_DSP_CMD_MASK (0xff) +#define RT5640_DSP_CMD_SFT 0 +#define RT5640_DSP_CMD_MW (0x3B) /* Memory Write */ +#define RT5640_DSP_CMD_MR (0x37) /* Memory Read */ +#define RT5640_DSP_CMD_RR (0x60) /* Register Read */ +#define RT5640_DSP_CMD_RW (0x68) /* Register Write */ + +/* Programmable Register Array Control 1 (0xc8) */ +#define RT5640_REG_SEQ_MASK (0xf << 12) +#define RT5640_REG_SEQ_SFT 12 +#define RT5640_SEQ1_ST_MASK (0x1 << 11) /*RO*/ +#define RT5640_SEQ1_ST_SFT 11 +#define RT5640_SEQ1_ST_RUN (0x0 << 11) +#define RT5640_SEQ1_ST_FIN (0x1 << 11) +#define RT5640_SEQ2_ST_MASK (0x1 << 10) /*RO*/ +#define RT5640_SEQ2_ST_SFT 10 +#define RT5640_SEQ2_ST_RUN (0x0 << 10) +#define RT5640_SEQ2_ST_FIN (0x1 << 10) +#define RT5640_REG_LV_MASK (0x1 << 9) +#define RT5640_REG_LV_SFT 9 +#define RT5640_REG_LV_MX (0x0 << 9) +#define RT5640_REG_LV_PR (0x1 << 9) +#define RT5640_SEQ_2_PT_MASK (0x1 << 8) +#define RT5640_SEQ_2_PT_BIT 8 +#define RT5640_REG_IDX_MASK (0xff) +#define RT5640_REG_IDX_SFT 0 + +/* Programmable Register Array Control 2 (0xc9) */ +#define RT5640_REG_DAT_MASK (0xffff) +#define RT5640_REG_DAT_SFT 0 + +/* Programmable Register Array Control 3 (0xca) */ +#define RT5640_SEQ_DLY_MASK (0xff << 8) +#define RT5640_SEQ_DLY_SFT 8 +#define RT5640_PROG_MASK (0x1 << 7) +#define RT5640_PROG_SFT 7 +#define RT5640_PROG_DIS (0x0 << 7) +#define RT5640_PROG_EN (0x1 << 7) +#define RT5640_SEQ1_PT_RUN (0x1 << 6) +#define RT5640_SEQ1_PT_RUN_BIT 6 +#define RT5640_SEQ2_PT_RUN (0x1 << 5) +#define RT5640_SEQ2_PT_RUN_BIT 5 + +/* Programmable Register Array Control 4 (0xcb) */ +#define RT5640_SEQ1_START_MASK (0xf << 8) +#define RT5640_SEQ1_START_SFT 8 +#define RT5640_SEQ1_END_MASK (0xf) +#define RT5640_SEQ1_END_SFT 0 + +/* Programmable Register Array Control 5 (0xcc) */ +#define RT5640_SEQ2_START_MASK (0xf << 8) +#define RT5640_SEQ2_START_SFT 8 +#define RT5640_SEQ2_END_MASK (0xf) +#define RT5640_SEQ2_END_SFT 0 + +/* Scramble Function (0xcd) */ +#define RT5640_SCB_KEY_MASK (0xff) +#define RT5640_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5640_SCB_SWAP_MASK (0x1 << 15) +#define RT5640_SCB_SWAP_SFT 15 +#define RT5640_SCB_SWAP_DIS (0x0 << 15) +#define RT5640_SCB_SWAP_EN (0x1 << 15) +#define RT5640_SCB_MASK (0x1 << 14) +#define RT5640_SCB_SFT 14 +#define RT5640_SCB_DIS (0x0 << 14) +#define RT5640_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5640_BB_MASK (0x1 << 15) +#define RT5640_BB_SFT 15 +#define RT5640_BB_DIS (0x0 << 15) +#define RT5640_BB_EN (0x1 << 15) +#define RT5640_BB_CT_MASK (0x7 << 12) +#define RT5640_BB_CT_SFT 12 +#define RT5640_BB_CT_A (0x0 << 12) +#define RT5640_BB_CT_B (0x1 << 12) +#define RT5640_BB_CT_C (0x2 << 12) +#define RT5640_BB_CT_D (0x3 << 12) +#define RT5640_M_BB_L_MASK (0x1 << 9) +#define RT5640_M_BB_L_SFT 9 +#define RT5640_M_BB_R_MASK (0x1 << 8) +#define RT5640_M_BB_R_SFT 8 +#define RT5640_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5640_M_BB_HPF_L_SFT 7 +#define RT5640_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5640_M_BB_HPF_R_SFT 6 +#define RT5640_G_BB_BST_MASK (0x3f) +#define RT5640_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5640_M_MP3_L_MASK (0x1 << 15) +#define RT5640_M_MP3_L_SFT 15 +#define RT5640_M_MP3_R_MASK (0x1 << 14) +#define RT5640_M_MP3_R_SFT 14 +#define RT5640_M_MP3_MASK (0x1 << 13) +#define RT5640_M_MP3_SFT 13 +#define RT5640_M_MP3_DIS (0x0 << 13) +#define RT5640_M_MP3_EN (0x1 << 13) +#define RT5640_EG_MP3_MASK (0x1f << 8) +#define RT5640_EG_MP3_SFT 8 +#define RT5640_MP3_HLP_MASK (0x1 << 7) +#define RT5640_MP3_HLP_SFT 7 +#define RT5640_MP3_HLP_DIS (0x0 << 7) +#define RT5640_MP3_HLP_EN (0x1 << 7) +#define RT5640_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5640_M_MP3_ORG_L_SFT 6 +#define RT5640_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5640_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5640_MP3_WT_MASK (0x1 << 13) +#define RT5640_MP3_WT_SFT 13 +#define RT5640_MP3_WT_1_4 (0x0 << 13) +#define RT5640_MP3_WT_1_2 (0x1 << 13) +#define RT5640_OG_MP3_MASK (0x1f << 8) +#define RT5640_OG_MP3_SFT 8 +#define RT5640_HG_MP3_MASK (0x3f) +#define RT5640_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5640_3D_CF_MASK (0x1 << 15) +#define RT5640_3D_CF_SFT 15 +#define RT5640_3D_CF_DIS (0x0 << 15) +#define RT5640_3D_CF_EN (0x1 << 15) +#define RT5640_3D_HP_MASK (0x1 << 14) +#define RT5640_3D_HP_SFT 14 +#define RT5640_3D_HP_DIS (0x0 << 14) +#define RT5640_3D_HP_EN (0x1 << 14) +#define RT5640_3D_BT_MASK (0x1 << 13) +#define RT5640_3D_BT_SFT 13 +#define RT5640_3D_BT_DIS (0x0 << 13) +#define RT5640_3D_BT_EN (0x1 << 13) +#define RT5640_3D_1F_MIX_MASK (0x3 << 11) +#define RT5640_3D_1F_MIX_SFT 11 +#define RT5640_3D_HP_M_MASK (0x1 << 10) +#define RT5640_3D_HP_M_SFT 10 +#define RT5640_3D_HP_M_SUR (0x0 << 10) +#define RT5640_3D_HP_M_FRO (0x1 << 10) +#define RT5640_M_3D_HRTF_MASK (0x1 << 9) +#define RT5640_M_3D_HRTF_SFT 9 +#define RT5640_M_3D_D2H_MASK (0x1 << 8) +#define RT5640_M_3D_D2H_SFT 8 +#define RT5640_M_3D_D2R_MASK (0x1 << 7) +#define RT5640_M_3D_D2R_SFT 7 +#define RT5640_M_3D_REVB_MASK (0x1 << 6) +#define RT5640_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5640_2ND_HPF_MASK (0x1 << 15) +#define RT5640_2ND_HPF_SFT 15 +#define RT5640_2ND_HPF_DIS (0x0 << 15) +#define RT5640_2ND_HPF_EN (0x1 << 15) +#define RT5640_HPF_CF_L_MASK (0x7 << 12) +#define RT5640_HPF_CF_L_SFT 12 +#define RT5640_1ST_HPF_MASK (0x1 << 11) +#define RT5640_1ST_HPF_SFT 11 +#define RT5640_1ST_HPF_DIS (0x0 << 11) +#define RT5640_1ST_HPF_EN (0x1 << 11) +#define RT5640_HPF_CF_R_MASK (0x7 << 8) +#define RT5640_HPF_CF_R_SFT 8 +#define RT5640_ZD_T_MASK (0x3 << 6) +#define RT5640_ZD_T_SFT 6 +#define RT5640_ZD_F_MASK (0x3 << 4) +#define RT5640_ZD_F_SFT 4 +#define RT5640_ZD_F_IM (0x0 << 4) +#define RT5640_ZD_F_ZC_IM (0x1 << 4) +#define RT5640_ZD_F_ZC_IOD (0x2 << 4) +#define RT5640_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5640_SI_DAC_MASK (0x1 << 11) +#define RT5640_SI_DAC_SFT 11 +#define RT5640_SI_DAC_AUTO (0x0 << 11) +#define RT5640_SI_DAC_TEST (0x1 << 11) +#define RT5640_DC_CAL_M_MASK (0x1 << 10) +#define RT5640_DC_CAL_M_SFT 10 +#define RT5640_DC_CAL_M_CAL (0x0 << 10) +#define RT5640_DC_CAL_M_NOR (0x1 << 10) +#define RT5640_DC_CAL_MASK (0x1 << 9) +#define RT5640_DC_CAL_SFT 9 +#define RT5640_DC_CAL_DIS (0x0 << 9) +#define RT5640_DC_CAL_EN (0x1 << 9) +#define RT5640_HPD_RCV_MASK (0x7 << 6) +#define RT5640_HPD_RCV_SFT 6 +#define RT5640_HPD_PS_MASK (0x1 << 5) +#define RT5640_HPD_PS_SFT 5 +#define RT5640_HPD_PS_DIS (0x0 << 5) +#define RT5640_HPD_PS_EN (0x1 << 5) +#define RT5640_CAL_M_MASK (0x1 << 4) +#define RT5640_CAL_M_SFT 4 +#define RT5640_CAL_M_DEP (0x0 << 4) +#define RT5640_CAL_M_CAL (0x1 << 4) +#define RT5640_CAL_MASK (0x1 << 3) +#define RT5640_CAL_SFT 3 +#define RT5640_CAL_DIS (0x0 << 3) +#define RT5640_CAL_EN (0x1 << 3) +#define RT5640_CAL_TEST_MASK (0x1 << 2) +#define RT5640_CAL_TEST_SFT 2 +#define RT5640_CAL_TEST_DIS (0x0 << 2) +#define RT5640_CAL_TEST_EN (0x1 << 2) +#define RT5640_CAL_P_MASK (0x3) +#define RT5640_CAL_P_SFT 0 +#define RT5640_CAL_P_NONE (0x0) +#define RT5640_CAL_P_CAL (0x1) +#define RT5640_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5640_SV_MASK (0x1 << 15) +#define RT5640_SV_SFT 15 +#define RT5640_SV_DIS (0x0 << 15) +#define RT5640_SV_EN (0x1 << 15) +#define RT5640_SPO_SV_MASK (0x1 << 14) +#define RT5640_SPO_SV_SFT 14 +#define RT5640_SPO_SV_DIS (0x0 << 14) +#define RT5640_SPO_SV_EN (0x1 << 14) +#define RT5640_OUT_SV_MASK (0x1 << 13) +#define RT5640_OUT_SV_SFT 13 +#define RT5640_OUT_SV_DIS (0x0 << 13) +#define RT5640_OUT_SV_EN (0x1 << 13) +#define RT5640_HP_SV_MASK (0x1 << 12) +#define RT5640_HP_SV_SFT 12 +#define RT5640_HP_SV_DIS (0x0 << 12) +#define RT5640_HP_SV_EN (0x1 << 12) +#define RT5640_ZCD_DIG_MASK (0x1 << 11) +#define RT5640_ZCD_DIG_SFT 11 +#define RT5640_ZCD_DIG_DIS (0x0 << 11) +#define RT5640_ZCD_DIG_EN (0x1 << 11) +#define RT5640_ZCD_MASK (0x1 << 10) +#define RT5640_ZCD_SFT 10 +#define RT5640_ZCD_PD (0x0 << 10) +#define RT5640_ZCD_PU (0x1 << 10) +#define RT5640_M_ZCD_MASK (0x3f << 4) +#define RT5640_M_ZCD_SFT 4 +#define RT5640_M_ZCD_RM_L (0x1 << 9) +#define RT5640_M_ZCD_RM_R (0x1 << 8) +#define RT5640_M_ZCD_SM_L (0x1 << 7) +#define RT5640_M_ZCD_SM_R (0x1 << 6) +#define RT5640_M_ZCD_OM_L (0x1 << 5) +#define RT5640_M_ZCD_OM_R (0x1 << 4) +#define RT5640_SV_DLY_MASK (0xf) +#define RT5640_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5640_ZCD_HP_MASK (0x1 << 15) +#define RT5640_ZCD_HP_SFT 15 +#define RT5640_ZCD_HP_DIS (0x0 << 15) +#define RT5640_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5640_3D_SPK_MASK (0x1 << 15) +#define RT5640_3D_SPK_SFT 15 +#define RT5640_3D_SPK_DIS (0x0 << 15) +#define RT5640_3D_SPK_EN (0x1 << 15) +#define RT5640_3D_SPK_M_MASK (0x3 << 13) +#define RT5640_3D_SPK_M_SFT 13 +#define RT5640_3D_SPK_CG_MASK (0x1f << 8) +#define RT5640_3D_SPK_CG_SFT 8 +#define RT5640_3D_SPK_SG_MASK (0x1f) +#define RT5640_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5640_WND_MASK (0x1 << 15) +#define RT5640_WND_SFT 15 +#define RT5640_WND_DIS (0x0 << 15) +#define RT5640_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5640_WND_FC_NW_MASK (0x3f << 10) +#define RT5640_WND_FC_NW_SFT 10 +#define RT5640_WND_FC_WK_MASK (0x3f << 4) +#define RT5640_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5640_HPF_FC_MASK (0x3f << 6) +#define RT5640_HPF_FC_SFT 6 +#define RT5640_WND_FC_ST_MASK (0x3f) +#define RT5640_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5640_WND_TH_LO_MASK (0x3ff) +#define RT5640_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5640_WND_TH_HI_MASK (0x3ff) +#define RT5640_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5640_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5640_WND_WIND_SFT 13 +#define RT5640_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5640_WND_STRONG_SFT 12 +enum { + RT5640_NO_WIND, + RT5640_BREEZE, + RT5640_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5640_DP_ATT_MASK (0x3 << 14) +#define RT5640_DP_ATT_SFT 14 +#define RT5640_DP_SPK_MASK (0x1 << 10) +#define RT5640_DP_SPK_SFT 10 +#define RT5640_DP_SPK_DIS (0x0 << 10) +#define RT5640_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5640_EQ_PRE_VOL_MASK (0xffff) +#define RT5640_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5640_EQ_PST_VOL_MASK (0xffff) +#define RT5640_EQ_PST_VOL_SFT 0 + +#define RT5640_NO_JACK BIT(0) +#define RT5640_HEADSET_DET BIT(1) +#define RT5640_HEADPHO_DET BIT(2) + +/* System Clock Source */ +#define RT5640_SCLK_S_MCLK 0 +#define RT5640_SCLK_S_PLL1 1 +#define RT5640_SCLK_S_PLL1_TK 2 +#define RT5640_SCLK_S_RCCLK 3 + +/* PLL1 Source */ +#define RT5640_PLL1_S_MCLK 0 +#define RT5640_PLL1_S_BCLK1 1 +#define RT5640_PLL1_S_BCLK2 2 +#define RT5640_PLL1_S_BCLK3 3 + + +enum { + RT5640_AIF1, + RT5640_AIF2, + RT5640_AIF3, + RT5640_AIFS, +}; + +enum { + RT5640_U_IF1 = 0x1, + RT5640_U_IF2 = 0x2, + RT5640_U_IF3 = 0x4, +}; + +enum { + RT5640_IF_123, + RT5640_IF_132, + RT5640_IF_312, + RT5640_IF_321, + RT5640_IF_231, + RT5640_IF_213, + RT5640_IF_113, + RT5640_IF_223, + RT5640_IF_ALL, +}; + +enum { + RT5640_DMIC_DIS, + RT5640_DMIC1, + RT5640_DMIC2, +}; + +struct rt5640_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5640_priv { + struct snd_soc_codec *codec; + struct rt5640_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5640_AIFS]; + int bclk[RT5640_AIFS]; + int master[RT5640_AIFS]; + + struct rt5640_pll_code pll_code; + int pll_src; + int pll_in; + int pll_out; + + int dmic_en; +}; + +#endif -- cgit v1.2.1 From 7c470373e097822ce6ca7bbac44b3afec0e7c1f8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 12 Jun 2013 17:44:08 +0100 Subject: ASoC: wm5102: Expose controls for DRE Certain use cases may require specific DRE settings so expose control of these. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 842adfdcb2a9..e723ac0c2113 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -816,6 +816,19 @@ SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), SOC_VALUE_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), +SOC_DOUBLE("HPOUT1 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0), +SOC_DOUBLE("HPOUT2 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0), +SOC_SINGLE("EPOUT DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE3L_ENA_SHIFT, 1, 0), + +SOC_SINGLE("DRE Threshold", ARIZONA_DRE_CONTROL_2, + ARIZONA_DRE_T_LOW_SHIFT, 63, 0), + +SOC_SINGLE("DRE Low Level ABS", ARIZONA_DRE_CONTROL_3, + ARIZONA_DRE_LOW_LEVEL_ABS_SHIFT, 15, 0), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- cgit v1.2.1 From dcad9f031240d59e9e1475a8e5b2cb427da94f6e Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 12 Jun 2013 11:34:30 -0600 Subject: ASoC: rt5640: add device tree support Modify the RT5640 driver to parse platform data from device tree. Write a DT binding document to describe those properties. Slight re-ordering of rt5640_i2c_probe() to better fit the DT parsing. Since ldo1_en is optional, guard usage of it with gpio_is_valid(), rather than open-coding an if (gpio) check. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 48 +++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 42 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 288c17cd6023..8761552882a5 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -1998,6 +1999,28 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) +{ + rt5640->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5640->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + + rt5640->pdata.ldo1_en = of_get_named_gpio(np, + "realtek,ldo1-en-gpios", 0); + /* + * LDO1_EN is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5640->pdata.ldo1_en) && + (rt5640->pdata.ldo1_en != -ENOENT)) + return rt5640->pdata.ldo1_en; + + return 0; +} + static int rt5640_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2011,6 +2034,24 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, GFP_KERNEL); if (NULL == rt5640) return -ENOMEM; + i2c_set_clientdata(i2c, rt5640); + + if (pdata) { + rt5640->pdata = *pdata; + /* + * Translate zero'd out (default) pdata value to an invalid + * GPIO ID. This makes the pdata and DT paths consistent in + * terms of the value left in this field when no GPIO is + * specified, but means we can't actually use GPIO 0. + */ + if (!rt5640->pdata.ldo1_en) + rt5640->pdata.ldo1_en = -EINVAL; + } else if (i2c->dev.of_node) { + ret = rt5640_parse_dt(rt5640, i2c->dev.of_node); + if (ret) + return ret; + } else + rt5640->pdata.ldo1_en = -EINVAL; rt5640->regmap = devm_regmap_init_i2c(i2c, &rt5640_regmap); if (IS_ERR(rt5640->regmap)) { @@ -2020,12 +2061,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, return ret; } - if (pdata) - rt5640->pdata = *pdata; - - i2c_set_clientdata(i2c, rt5640); - - if (rt5640->pdata.ldo1_en) { + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { ret = devm_gpio_request_one(&i2c->dev, rt5640->pdata.ldo1_en, GPIOF_OUT_INIT_HIGH, "RT5640 LDO1_EN"); -- cgit v1.2.1 From 040a62cf1c040362fb11587fb9f02e1881f4c237 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 12 Jun 2013 11:35:34 -0600 Subject: ASoC: tegra: add tegra+RT5640 machine driver Initially, this binding and driver only describe/support playback to headphones and speakers. This driver will support Beaver and Dalmore. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 10 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5640.c | 257 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 269 insertions(+) create mode 100644 sound/soc/tegra/tegra_rt5640.c (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index b1c9d573da05..995b120c2cd0 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -59,6 +59,16 @@ config SND_SOC_TEGRA30_I2S Tegra30 I2S interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA_RT5640 + tristate "SoC Audio support for Tegra boards using an RT5640 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5640 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5640 codec, such as Dalmore. + config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 416a14bde41b..21d2550a08a4 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -18,12 +18,14 @@ obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support +snd-soc-tegra-rt5640-objs := tegra_rt5640.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c new file mode 100644 index 000000000000..08794f915a94 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5640.c @@ -0,0 +1,257 @@ +/* +* tegra_rt5640.c - Tegra machine ASoC driver for boards using WM8903 codec. + * + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "../codecs/rt5640.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5640" + +struct tegra_rt5640 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; +}; + +static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_rt5640_ops = { + .hw_params = tegra_rt5640_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5640_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5640_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, +}; + +static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5640_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), +}; + +static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card); + + snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins), + tegra_rt5640_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5640_hp_jack, + 1, + &tegra_rt5640_hp_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5640_dai = { + .name = "RT5640", + .stream_name = "RT5640 PCM", + .codec_dai_name = "rt5640-aif1", + .init = tegra_rt5640_asoc_init, + .ops = &tegra_rt5640_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5640 = { + .name = "tegra-rt5640", + .owner = THIS_MODULE, + .dai_link = &tegra_rt5640_dai, + .num_links = 1, + .controls = tegra_rt5640_controls, + .num_controls = ARRAY_SIZE(tegra_rt5640_controls), + .dapm_widgets = tegra_rt5640_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5640_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5640_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5640; + struct tegra_rt5640 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5640), GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_rt5640\n"); + return -ENOMEM; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_rt5640_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5640_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5640_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5640_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5640_dai.platform_of_node = tegra_rt5640_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5640_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1, + &tegra_rt5640_hp_jack_gpio); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5640_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5640", }, + {}, +}; + +static struct platform_driver tegra_rt5640_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5640_of_match, + }, + .probe = tegra_rt5640_probe, + .remove = tegra_rt5640_remove, +}; +module_platform_driver(tegra_rt5640_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra+RT5640 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5640_of_match); -- cgit v1.2.1 From d8f4e17fdd4f5e6fe6ef07496296fa88e150beda Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 12 Jun 2013 22:18:01 +0200 Subject: ASoC: SPEAr spdif_{in,out}: fix fallout of previous cleanup MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The patch that resulted in bfcc74e (ASoC: SPEAr spdif_{in,out}: use devm for clk and a few more cleanups) was broken and applied on a newer tree than it was created for. So bfcc74e introduced unbalanced clk handling, two warnings about unused variables and passed 3 arguments to a function only taking 2. This commit fixes that. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/spear/spdif_in.c | 10 +--------- sound/soc/spear/spdif_out.c | 12 ++---------- 2 files changed, 3 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index f0071ddbfa7d..63acfeb4b69d 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -257,20 +257,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_register_component(&pdev->dev, &spdif_in_component, + return snd_soc_register_component(&pdev->dev, &spdif_in_component, &spdif_in_dai, 1); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; } static int spdif_in_remove(struct platform_device *pdev) { - struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 4bde5123cea6..a4a874820ab1 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -298,8 +298,7 @@ static int spdif_out_probe(struct platform_device *pdev) return -ENOMEM; } - host->io_base = devm_request_and_ioremap(&pdev->dev, res->start, - resource_size(res)); + host->io_base = devm_request_and_ioremap(&pdev->dev, res); if (!host->io_base) { dev_warn(&pdev->dev, "ioremap failed\n"); return -ENOMEM; @@ -321,18 +320,11 @@ static int spdif_out_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &spdif_out_component, &spdif_out_dai, 1); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; + return ret; } static int spdif_out_remove(struct platform_device *pdev) { - struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; -- cgit v1.2.1 From 9be94aeabf551aa5e2481ab9d626ba82d6b14f3d Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 12 Jun 2013 15:34:23 -0600 Subject: ASoC: rt5640: fix sparse warnings This fixes: 975:9: sparse: Using plain integer as NULL pointer 1917:24: sparse: symbol 'rt5640_aif_dai_ops' was not declared. Should it be static? 1924:27: sparse: symbol 'rt5640_dai' was not declared. Should it be static? 2079:19: sparse: symbol 'rt5640_i2c_driver' was not declared. Should it be static? Reported-by: Fengguang Wu Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8761552882a5..ce585e37e38a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -974,7 +974,7 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5640_PWR_ANLG1, RT5640_PWR_LDO2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5640_PWR_ANLG2, - RT5640_PWR_MB1_BIT, 0, 0, 0), + RT5640_PWR_MB1_BIT, 0, NULL, 0), /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1"), SND_SOC_DAPM_INPUT("DMIC2"), @@ -1915,14 +1915,14 @@ static int rt5640_resume(struct snd_soc_codec *codec) #define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5640_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5640_aif_dai_ops = { .hw_params = rt5640_hw_params, .set_fmt = rt5640_set_dai_fmt, .set_sysclk = rt5640_set_dai_sysclk, .set_pll = rt5640_set_dai_pll, }; -struct snd_soc_dai_driver rt5640_dai[] = { +static struct snd_soc_dai_driver rt5640_dai[] = { { .name = "rt5640-aif1", .id = RT5640_AIF1, @@ -2112,7 +2112,7 @@ static int rt5640_i2c_remove(struct i2c_client *i2c) return 0; } -struct i2c_driver rt5640_i2c_driver = { +static struct i2c_driver rt5640_i2c_driver = { .driver = { .name = "rt5640", .owner = THIS_MODULE, -- cgit v1.2.1 From 2e7ee15ced914e109a1a5b6dfcd463d846a13bd5 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 14 Jun 2013 12:34:50 +0800 Subject: ASoC: wm8962: Remove remaining direct register cache accesses Also fix return values for headphone switch updates. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 24 +++++++++++++----------- 1 file changed, 13 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e9710280e5e1..730dd0c0f0ab 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1600,7 +1600,6 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -1609,16 +1608,19 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, - reg_cache[WM8962_HPOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_HPOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_HPOUTL_VOLUME, + snd_soc_read(codec, WM8962_HPOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, - reg_cache[WM8962_HPOUTR_VOLUME]); + if (ret & WM8962_HPOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_HPOUTR_VOLUME, + snd_soc_read(codec, WM8962_HPOUTR_VOLUME)); - return 0; + return 1; } /* The VU bits for the speakers are in a different register to the mute @@ -3374,7 +3376,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); - u16 *reg_cache = codec->reg_cache; int i, trigger, irq_pol; bool dmicclk, dmicdat; @@ -3432,8 +3433,9 @@ static int wm8962_probe(struct snd_soc_codec *codec) /* Put the speakers into mono mode? */ if (pdata->spk_mono) - reg_cache[WM8962_CLASS_D_CONTROL_2] - |= WM8962_SPK_MONO; + snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); + /* Micbias setup, detection enable and detection * threasholds. */ -- cgit v1.2.1 From 55dafe5dc29c1a8e9ac4a14c38f9550ec79524c9 Mon Sep 17 00:00:00 2001 From: Emil Goode Date: Fri, 14 Jun 2013 11:48:56 +0200 Subject: ASoC: Fix double assignment of .owner in struct snd_soc_card In struct snd_soc_card zylonite .owner is assigned THIS_MODULE twice, remove one of them. Signed-off-by: Emil Goode Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ceb656695b0f..db8aadf8932d 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -256,7 +256,6 @@ static struct snd_soc_card zylonite = { .resume_pre = &zylonite_resume_pre, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), - .owner = THIS_MODULE, }; static struct platform_device *zylonite_snd_ac97_device; -- cgit v1.2.1 From 6df2610c150c3e7088ac9b82df3e9ee732802409 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 14 Jun 2013 12:26:20 +0100 Subject: ASoC: SPEAr: Add dependency on dmaengine helpers I'd be very surprised if anyone has used the Kconfig... Signed-off-by: Mark Brown --- sound/soc/spear/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index 3b7cdadc11cc..5b744ed52988 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -27,6 +27,7 @@ config SND_SPEAR_SOC Say Y or M if you want to add support for any of the audio controllers (I2S/SPDIF). You will also need to select the audio interface codecs to support below. + select SND_SOC_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT tristate "SPEAr SPDIF Out Device Driver" -- cgit v1.2.1 From 41139938cd63e9bf1a1eda62cb2add8dd8b561b4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 14 Jun 2013 12:28:46 +0100 Subject: ASoC: SPEAr: Hide component drivers in Kconfig The individual component drivers are only useful with a machine driver and should be selected by the machine drivers so shouldn't have help text of their own in order to hide them in interactive configuration. Signed-off-by: Mark Brown --- sound/soc/spear/Kconfig | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index 5b744ed52988..e3851d67bfc4 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -20,21 +20,11 @@ config SND_SPEAR1340_EVM platform config SND_SPEAR_SOC - tristate "SoC Audio for the ST chip" - depends on SND_DESIGNWARE_I2S || SND_SPEAR_SPDIF_OUT || \ - SND_SPEAR_SPDIF_IN - help - Say Y or M if you want to add support for any of the audio - controllers (I2S/SPDIF). You will also need to select - the audio interface codecs to support below. + tristate select SND_SOC_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT - tristate "SPEAr SPDIF Out Device Driver" - help - Say Y or M if you want to add support for SPDIF OUT driver. + tristate config SND_SPEAR_SPDIF_IN - tristate "SPEAr SPDIF IN Device Driver" - help - Say Y or M if you want to add support for SPDIF IN driver. + tristate -- cgit v1.2.1 From 6b75bf0c5b17d71c3b0568caaaaae73a22f83826 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Jun 2013 13:16:51 +0200 Subject: ASoC: dapm: Setup private_free callback for dapm kcontrols The private data containing the widget list that is a assigned to a DAPM kcontrol is never freed. Setup the private_free for DAPM kcontrols to take care of this. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a80c883bb8be..9dd2d1d63981 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -504,6 +504,11 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, return 0; } +static void dapm_kcontrol_free(struct snd_kcontrol *kctl) +{ + kfree(kctl->private_data); +} + /* * Determine if a kcontrol is shared. If it is, look it up. If it isn't, * create it. Either way, add the widget into the control's widget list @@ -617,6 +622,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name, prefix); + kcontrol->private_free = dapm_kcontrol_free; ret = snd_ctl_add(card, kcontrol); if (ret < 0) { dev_err(dapm->dev, -- cgit v1.2.1 From 58fee775b7a18a0174931af6174536560785d500 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Jun 2013 13:16:52 +0200 Subject: ASoC: dapm: Remove unnecessary loop The condition 'i == item' is only true when, well, 'i' equals 'item'. So just use 'item' directly as the index into the array. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9dd2d1d63981..8d8a8dc6857e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -363,11 +363,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, val = soc_widget_read(w, e->reg); item = (val >> e->shift_l) & e->mask; - p->connect = 0; - for (i = 0; i < e->max; i++) { - if (!(strcmp(p->name, e->texts[i])) && item == i) - p->connect = 1; - } + if (item < e->max && !strcmp(p->name, e->texts[item])) + p->connect = 1; + else + p->connect = 0; } break; case snd_soc_dapm_virt_mux: { @@ -397,11 +396,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, break; } - p->connect = 0; - for (i = 0; i < e->max; i++) { - if (!(strcmp(p->name, e->texts[i])) && item == i) - p->connect = 1; - } + if (item < e->max && !strcmp(p->name, e->texts[item])) + p->connect = 1; + else + p->connect = 0; } break; /* does not affect routing - always connected */ -- cgit v1.2.1 From 8872293fc38c4906c86e7d335b8f936abf9e4531 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Jun 2013 13:16:53 +0200 Subject: ASoC: dapm: Add a helper function to free a DAPM path We have the same code for freeing a DAPM path in three different locations. Introduce a new helper function to take care of this. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 34 ++++++++++++++++------------------ 1 file changed, 16 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8d8a8dc6857e..784534dcc82d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2105,6 +2105,15 @@ static void snd_soc_dapm_sys_remove(struct device *dev) device_remove_file(dev, &dev_attr_dapm_widget); } +static void dapm_free_path(struct snd_soc_dapm_path *path) +{ + list_del(&path->list_sink); + list_del(&path->list_source); + list_del(&path->list); + kfree(path->long_name); + kfree(path); +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2120,20 +2129,12 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) * While removing the path, remove reference to it from both * source and sink widgets so that path is removed only once. */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p->long_name); - kfree(p); - } - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p->long_name); - kfree(p); - } + list_for_each_entry_safe(p, next_p, &w->sources, list_sink) + dapm_free_path(p); + + list_for_each_entry_safe(p, next_p, &w->sinks, list_source) + dapm_free_path(p); + kfree(w->kcontrols); kfree(w->name); kfree(w); @@ -2408,10 +2409,7 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(path->source, "Route removed"); dapm_mark_dirty(path->sink, "Route removed"); - list_del(&path->list); - list_del(&path->list_sink); - list_del(&path->list_source); - kfree(path); + dapm_free_path(path); } else { dev_warn(dapm->dev, "ASoC: Route %s->%s does not exist\n", source, sink); -- cgit v1.2.1 From 656ca9d327a3dbac6db28c5bf80f5bc86f7f8548 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Jun 2013 13:16:54 +0200 Subject: ASoC: dapm: Remove unused long_name field from snd_soc_dapm_path struct Since commit 85762e71 ("ASoC: dapm: Implement mixer control sharing") the long_name field of the snd_soc_dapm_path struct is unused. All of the name handling now happens entirely in dapm_create_or_share_mixmux_kcontrol(). So we can remove the long_name field from the snd_soc_dapm_path struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 784534dcc82d..163f26d9571c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -621,17 +621,15 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name, prefix); kcontrol->private_free = dapm_kcontrol_free; + kfree(long_name); ret = snd_ctl_add(card, kcontrol); if (ret < 0) { dev_err(dapm->dev, "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", w->name, name, ret); kfree(wlist); - kfree(long_name); return ret; } - - path->long_name = long_name; } kcontrol->private_data = wlist; @@ -2110,7 +2108,6 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) list_del(&path->list_sink); list_del(&path->list_source); list_del(&path->list); - kfree(path->long_name); kfree(path); } -- cgit v1.2.1 From efc77e36ae6ff4394a0232a4f87bded0bd555d6b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Jun 2013 13:16:50 +0200 Subject: ASoC: dapm: Add snd_soc_dapm_switch to the power up/down sequence table The power up/down sequence order for DAPM switch widgets is not explicitly initialized, causing them to be run always as the first widget type for both power up and down. Move it to the same position in the sequence as other mixer widget types. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a80c883bb8be..2324f3cfa0d0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -63,6 +63,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_virt_mux] = 5, [snd_soc_dapm_value_mux] = 5, [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_switch] = 7, [snd_soc_dapm_mixer] = 7, [snd_soc_dapm_mixer_named_ctl] = 7, [snd_soc_dapm_pga] = 8, @@ -82,6 +83,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, -- cgit v1.2.1 From f5055f93733730b61a8a69dedbb216e6b4dd84c5 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 14 Jun 2013 19:49:06 +0800 Subject: ASoC: wm8962: Enable start-up and normal bias after reset in runtime resume This part of bias settings are essential for WM8962 to power up. Without it "wm8962 0-001a: DC servo timed out" might be prompted due to power-up failure that happens to FLL if being used. The driver's also bringing the bias down in the suspend path so it needs to be powered up in the resume path for symmetry. According to dapm_pre_sequence_async(), DAPM would call pm_runtime_get_sync() to let driver finish the bias settings in pm_runtime_resume() before the bias level being set to STANDBY. So no need to worry about disordered settings for VMID of WM8962. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 730dd0c0f0ab..4b7915bec2f1 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3723,6 +3723,17 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); + + /* Bias enable at 2*5k (fast start-up) */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_BIAS_ENA | WM8962_VMID_SEL_MASK, + WM8962_BIAS_ENA | 0x180); + + msleep(5); + return 0; } -- cgit v1.2.1 From 5be9c5b477bd3eda6bbe0f6e3431da45c3fd28f4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 14 Jun 2013 14:19:36 +0100 Subject: ASoC: wm5110: Correct rate control for DSP4 Reported-by: Dennis May Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b649c0b2..0a3c7f704016 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -279,7 +279,7 @@ static const struct soc_enum wm_adsp2_rate_enum[] = { ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, arizona_rate_text, arizona_rate_val), -- cgit v1.2.1 From fc7fe01518b0a29750313a7cc9dc8f8a0416a6b4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Jun 2013 08:58:52 +0100 Subject: ASoC: spear: Remove nonexistant EVM options The source wasn't added. Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/spear/Kconfig | 21 --------------------- sound/soc/spear/Makefile | 3 --- 2 files changed, 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index e3851d67bfc4..3567d73b218e 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,24 +1,3 @@ -config SND_SPEAR_EVM - tristate "SoC Audio support for SPEAr EVM" - select SND_DESIGNWARE_I2S - select SND_SOC_STA529 - select SND_SPEAR_SOC - help - Say Y if you want to add support for SoC audio on SPEAr - platform - -config SND_SPEAR1340_EVM - tristate "SoC Audio support for SPEAr1340 EVM" - select SND_DESIGNWARE_I2S - select SND_SOC_STA529 - select SND_SPEAR_SPDIF_OUT - select SND_SPEAR_SPDIF_IN - select SND_SOC_SPDIF - select SND_SPEAR_SOC - help - Say Y if you want to add support for SoC audio on SPEAr1340 - platform - config SND_SPEAR_SOC tristate select SND_SOC_DMAENGINE_PCM diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile index b36512655bcf..9f47bb428601 100644 --- a/sound/soc/spear/Makefile +++ b/sound/soc/spear/Makefile @@ -2,6 +2,3 @@ obj-$(CONFIG_SND_SPEAR_SOC) += spear_pcm.o obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += spdif_in.o obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += spdif_out.o - -# SPEAR Machine Support -obj-$(CONFIG_SND_SPEAR_EVM) += spear_evb.o -- cgit v1.2.1 From da177dd0259eb74061a3a1f6ab037f5080c2a504 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 9 Jun 2013 16:14:25 +0300 Subject: ALSA: usx2y: remove some old dead code USB_QUEUE_BULK isn't defined any more. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2yaudio.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index b37653247ef4..4967fe9c938d 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -695,9 +695,6 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) ((char*)(usbdata + i))[1] = ra[i].c2; usb_fill_bulk_urb(us->urb[i], usX2Y->dev, usb_sndbulkpipe(usX2Y->dev, 4), usbdata + i, 2, i_usX2Y_04Int, usX2Y); -#ifdef OLD_USB - us->urb[i]->transfer_flags = USB_QUEUE_BULK; -#endif } us->submitted = 0; us->len = NOOF_SETRATE_URBS; -- cgit v1.2.1 From 4867e99d2165416f8b1093c3267b5f5ac5ebf8ef Mon Sep 17 00:00:00 2001 From: Sekhar Nori Date: Mon, 17 Jun 2013 14:16:31 +0530 Subject: ASoC: davinci: remove sffsdr machine support sffsdr machine support does not build since at least v2.6.36 (~3 years). There is little hope of it being fixed, so remove the support. Signed-off-by: Sekhar Nori Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 10 -- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-sffsdr.c | 181 ------------------------------------- 3 files changed, 193 deletions(-) delete mode 100644 sound/soc/davinci/davinci-sffsdr.c (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 9e11a14d1b45..c82f89c9475b 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -54,16 +54,6 @@ config SND_DM6467_SOC_EVM help Say Y if you want to add support for SoC audio on TI -config SND_DAVINCI_SOC_SFFSDR - tristate "SoC Audio support for SFFSDR" - depends on SND_DAVINCI_SOC && MACH_SFFSDR - select SND_DAVINCI_SOC_I2S - select SND_SOC_PCM3008 - select SFFSDR_FPGA - help - Say Y if you want to add support for SoC audio on - Lyrtech SFFSDR board. - config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a93679d618cd..a396ab6d6d5e 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -11,10 +11,8 @@ obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o -snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c deleted file mode 100644 index 24df5bcf3bbc..000000000000 --- a/sound/soc/davinci/davinci-sffsdr.c +++ /dev/null @@ -1,181 +0,0 @@ -/* - * ASoC driver for Lyrtech SFFSDR board. - * - * Author: Hugo Villeneuve - * Copyright (C) 2008 Lyrtech inc - * - * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: - * Copyright: (C) 2007 MontaVista Software, Inc., - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#ifdef CONFIG_SFFSDR_FPGA -#include -#endif - -#include - -#include "../codecs/pcm3008.h" -#include "davinci-pcm.h" -#include "davinci-i2s.h" - -/* - * CLKX and CLKR are the inputs for the Sample Rate Generator. - * FSX and FSR are outputs, driven by the sample Rate Generator. - */ -#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ - SND_SOC_DAIFMT_CBM_CFS | \ - SND_SOC_DAIFMT_IB_NF) - -static int sffsdr_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int fs; - int ret = 0; - - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - -#ifndef CONFIG_SFFSDR_FPGA - /* Without the FPGA module, the Fs is fixed at 44100 Hz */ - if (fs != 44100) { - pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); - return -EINVAL; - } -#endif - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; - - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); - -#ifndef CONFIG_SFFSDR_FPGA - return 0; -#else - return sffsdr_fpga_set_codec_fs(fs); -#endif -} - -static struct snd_soc_ops sffsdr_ops = { - .hw_params = sffsdr_hw_params, -}; - -/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sffsdr_dai = { - .name = "PCM3008", /* Codec name */ - .stream_name = "PCM3008 HiFi", - .cpu_dai_name = "davinci-mcbsp", - .codec_dai_name = "pcm3008-hifi", - .codec_name = "pcm3008-codec", - .platform_name = "davinci-mcbsp", - .ops = &sffsdr_ops, -}; - -/* davinci-sffsdr audio machine driver */ -static struct snd_soc_card snd_soc_sffsdr = { - .name = "DaVinci SFFSDR", - .owner = THIS_MODULE, - .dai_link = &sffsdr_dai, - .num_links = 1, -}; - -/* sffsdr audio private data */ -static struct pcm3008_setup_data sffsdr_pcm3008_setup = { - .dem0_pin = GPIO(45), - .dem1_pin = GPIO(46), - .pdad_pin = GPIO(47), - .pdda_pin = GPIO(38), -}; - -static struct platform_device pcm3008_codec = { - .name = "pcm3008-codec", - .id = 0, - .dev = { - .platform_data = &sffsdr_pcm3008_setup, - }, -}; - -static struct resource sffsdr_snd_resources[] = { - { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data sffsdr_snd_data = { - .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, - .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, -}; - -static struct platform_device *sffsdr_snd_device; - -static int __init sffsdr_init(void) -{ - int ret; - - if (!machine_is_sffsdr()) - return -EINVAL; - - platform_device_register(&pcm3008_codec); - - sffsdr_snd_device = platform_device_alloc("soc-audio", 0); - if (!sffsdr_snd_device) { - printk(KERN_ERR "platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(sffsdr_snd_device, &snd_soc_sffsdr); - platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data, - sizeof(sffsdr_snd_data)); - - ret = platform_device_add_resources(sffsdr_snd_device, - sffsdr_snd_resources, - ARRAY_SIZE(sffsdr_snd_resources)); - if (ret) { - printk(KERN_ERR "platform device add resources failed\n"); - goto error; - } - - ret = platform_device_add(sffsdr_snd_device); - if (ret) - goto error; - - return ret; - -error: - platform_device_put(sffsdr_snd_device); - return ret; -} - -static void __exit sffsdr_exit(void) -{ - platform_device_unregister(sffsdr_snd_device); - platform_device_unregister(&pcm3008_codec); -} - -module_init(sffsdr_init); -module_exit(sffsdr_exit); - -MODULE_AUTHOR("Hugo Villeneuve"); -MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); -MODULE_LICENSE("GPL"); -- cgit v1.2.1 From 1d26f752ace99c30ceadb4e5c106736b15da99e1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Jun 2013 09:02:11 +0100 Subject: ASoC: spear: Normalise module names Signed-off-by: Mark Brown --- sound/soc/spear/Makefile | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile index 9f47bb428601..c4ea7161056c 100644 --- a/sound/soc/spear/Makefile +++ b/sound/soc/spear/Makefile @@ -1,4 +1,8 @@ # SPEAR Platform Support -obj-$(CONFIG_SND_SPEAR_SOC) += spear_pcm.o -obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += spdif_in.o -obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += spdif_out.o +snd-soc-spear-pcm-objs := spear_pcm.o +snd-soc-spear-spdif-in-objs := spdif_in.o +snd-soc-spear-spdif-out-objs := spdif_out.o + +obj-$(CONFIG_SND_SPEAR_SOC) += snd-soc-spear-pcm.o +obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += snd-soc-spear-spdif-in.o +obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += snd-soc-spear-spdif-out.o -- cgit v1.2.1 From fc09cfbe3e3535897456c12f37fa83024bdab92d Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Mon, 17 Jun 2013 16:10:57 +0530 Subject: ASoC: spear: Convert to use devm_ioremap_resource Commit 75096579c3ac ("lib: devres: Introduce devm_ioremap_resource()") introduced devm_ioremap_resource() and deprecated the use of devm_request_and_ioremap(). devm_request_mem_region is called in devm_ioremap_resource(). Hence that part can also be removed. Since devm_ioremap_resource prints error message on failure, there is no need to print an explicit warning message. Signed-off-by: Tushar Behera CC: alsa-devel@alsa-project.org CC: Liam Girdwood CC: Mark Brown Signed-off-by: Mark Brown --- sound/soc/spear/spdif_out.c | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index a4a874820ab1..2fdf68c98d22 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -282,27 +282,16 @@ static int spdif_out_probe(struct platform_device *pdev) struct resource *res; int ret; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -EINVAL; - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_warn(&pdev->dev, "Failed to get memory resourse\n"); - return -ENOENT; - } - host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - host->io_base = devm_request_and_ioremap(&pdev->dev, res); - if (!host->io_base) { - dev_warn(&pdev->dev, "ioremap failed\n"); - return -ENOMEM; - } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + host->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(host->io_base)) + return PTR_ERR(host->io_base); host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) -- cgit v1.2.1 From cd1199edc719f4a918a19bd2c6b8f79329837561 Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Mon, 17 Jun 2013 22:26:57 -0400 Subject: ALSA: sound/usb/misc/ua101.c: convert __list_for_each usage to list_for_each Signed-off-by: Dave Jones Signed-off-by: Takashi Iwai --- sound/usb/misc/ua101.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 6ad617b94732..8b5d2c564e04 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1349,7 +1349,7 @@ static void ua101_disconnect(struct usb_interface *interface) snd_card_disconnect(ua->card); /* make sure that there are no pending USB requests */ - __list_for_each(midi, &ua->midi_list) + list_for_each(midi, &ua->midi_list) snd_usbmidi_disconnect(midi); abort_alsa_playback(ua); abort_alsa_capture(ua); -- cgit v1.2.1 From 06ec56d3c60238f27bfa50d245592fccc1b4ef0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Jun 2013 07:55:02 +0200 Subject: ALSA: hda - Fix return value of snd_hda_check_power_state() The refactoring by commit 9040d102 introduced the new function snd_hda_check_power_state(). This function is supposed to return true if the state already reached to the target state, but it actually returns false for that. An utterly stupid typo while copy & paste. Fortunately this didn't influence on much behavior because powering up AFG usually powers up the child widgets, too. But the finer power control must have been broken by this bug. Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e0bf7534fa1f..29ed7d9b27e4 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -667,7 +667,7 @@ snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid, if (state & AC_PWRST_ERROR) return true; state = (state >> 4) & 0x0f; - return (state != target_state); + return (state == target_state); } unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, -- cgit v1.2.1 From 53b434f09340db8ad59b43789b7c43f54171fe36 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Tue, 18 Jun 2013 10:41:53 +0800 Subject: ALSA: hda - Haswell converter power state D0 verify Haswell converters maybe in wrong power state before usage. i.e. only converter 0 is in D0, converter 1/2 are in D3. When pin choose converter 1/2, there's no audio output, this cause dependency when playing differnt stream on pins. AUD_PWRST ConvertorA_Widget_Power_State_Current D0 AUD_PWRST ConvertorA_Widget_Power_State_Requsted D0 AUD_PWRST ConvertorB_Widget_Power_State_Current D3 AUD_PWRST ConvertorB_Widget_Power_State_Requested D3 AUD_PWRST ConvC_Widget_PwrSt_Curr D3 AUD_PWRST ConvC_Widget_PwrSt_Req D3 This patch check converter's power state and set D0 if it's in D3 mode. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a030c58..8983747f2a37 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1018,10 +1018,19 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, hda_nid_t nid) +static void haswell_verify_pin_D0(struct hda_codec *codec, + hda_nid_t cvt_nid, hda_nid_t nid) { int pwr, lamp, ramp; + /* For Haswell, the converter 1/2 may keep in D3 state after bootup, + * thus pins could only choose converter 0 for use. Make sure the + * converters are in correct power state */ + pwr = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_POWER_STATE, 0); + pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; + if (pwr != AC_PWRST_D0) + snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; if (pwr != AC_PWRST_D0) { @@ -1068,7 +1077,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int new_pinctl = 0; if (codec->vendor_id == 0x80862807) - haswell_verify_pin_D0(codec, pin_nid); + haswell_verify_pin_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, -- cgit v1.2.1 From 1389fd03b7ff72625cdae5cc3f838ce093661200 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 18 Jun 2013 21:09:42 +0800 Subject: ALSA: firewire: fix error return code in scs_probe() Fix to return -ENOMEM in the kmalloc() error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/scs1x.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/scs1x.c b/sound/firewire/scs1x.c index 844a555c3b1e..b252c21b6d13 100644 --- a/sound/firewire/scs1x.c +++ b/sound/firewire/scs1x.c @@ -405,8 +405,10 @@ static int scs_probe(struct device *unit_dev) scs->output_idle = true; scs->buffer = kmalloc(HSS1394_MAX_PACKET_SIZE, GFP_KERNEL); - if (!scs->buffer) + if (!scs->buffer) { + err = -ENOMEM; goto err_card; + } scs->hss_handler.length = HSS1394_MAX_PACKET_SIZE; scs->hss_handler.address_callback = handle_hss; -- cgit v1.2.1 From bddee96b5d0db869f47b195fe48c614ca824203c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Jun 2013 16:14:22 +0200 Subject: ALSA: hda - Cache the MUX selection for generic HDMI When a selection to a converter MUX is changed in hdmi_pcm_open(), it should be cached so that the given connection can be restored properly at PM resume. We need just to replace the corresponding snd_hda_codec_write() call with snd_hda_codec_write_cache(). Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8983747f2a37..844cf55a62b1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1155,7 +1155,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - snd_hda_codec_write(codec, per_pin->pin_nid, 0, + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); -- cgit v1.2.1 From 7ef166b831237e67b2ea83ce0c933c46ddd6eb26 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Tue, 18 Jun 2013 21:42:14 +0800 Subject: ALSA: hda - Avoid choose same converter for unused pins For Intel Haswell HDMI codecs, the pins choose converter 0 by default. This would cause conflict when playing audio on unused pins,the pin with physical device connected would get audio data too. i.e. Pin 0/1/2 default choose converter 0, pin 1 has HDMI monitor connected. when play audio on Pin 0 or pin 2, pin 1 could get audio data too. This patch configure unused pins to choose different converter. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 89 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 75 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 844cf55a62b1..0687d536b563 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1110,26 +1110,15 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return 0; } -/* - * HDA PCM callbacks - */ -static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int hdmi_choose_cvt(struct hda_codec *codec, + int pin_idx, int *cvt_id, int *mux_id) { struct hdmi_spec *spec = codec->spec; - struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, mux_idx = 0; struct hdmi_spec_per_pin *per_pin; - struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; + int cvt_idx, mux_idx = 0; - /* Validate hinfo */ - pin_idx = hinfo_to_pin_index(spec, hinfo); - if (snd_BUG_ON(pin_idx < 0)) - return -EINVAL; per_pin = get_pin(spec, pin_idx); - eld = &per_pin->sink_eld; /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { @@ -1147,10 +1136,77 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, continue; break; } + /* No free converters */ if (cvt_idx == spec->num_cvts) return -ENODEV; + if (cvt_id) + *cvt_id = cvt_idx; + if (mux_id) + *mux_id = mux_idx; + + return 0; +} + +static void haswell_config_cvts(struct hda_codec *codec, + int pin_id, int mux_id) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; + int pin_idx, mux_idx; + int curr; + int err; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + per_pin = get_pin(spec, pin_idx); + + if (pin_idx == pin_id) + continue; + + curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + + /* Choose another unused converter */ + if (curr == mux_id) { + err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); + if (err < 0) + return; + snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); + } + } +} + +/* + * HDA PCM callbacks + */ +static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int pin_idx, cvt_idx, mux_idx = 0; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + struct hdmi_spec_per_cvt *per_cvt = NULL; + int err; + + /* Validate hinfo */ + pin_idx = hinfo_to_pin_index(spec, hinfo); + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; + + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + if (err < 0) + return err; + + per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; @@ -1158,6 +1214,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); + + /* configure unused pins to choose other converters */ + if (codec->vendor_id == 0x80862807) + haswell_config_cvts(codec, pin_idx, mux_idx); + snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ -- cgit v1.2.1 From fd678cac34e66b5a289e1abd159c3cb080040370 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Jun 2013 16:28:36 +0200 Subject: ALSA: hda - Use snd_hda_check_power_state() in patch_hdmi.c ... instead of open codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0687d536b563..49ef8f8eb5e9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1026,14 +1026,10 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, /* For Haswell, the converter 1/2 may keep in D3 state after bootup, * thus pins could only choose converter 0 for use. Make sure the * converters are in correct power state */ - pwr = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_POWER_STATE, 0); - pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; - if (pwr != AC_PWRST_D0) + if (!snd_hda_check_power_state(codec, cvt_nid, AC_PWRST_D0)) snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); - pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; - if (pwr != AC_PWRST_D0) { + if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D0)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); msleep(40); -- cgit v1.2.1 From d045c5dc43d829df9f067d363c3b42b14dacf434 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 19 Jun 2013 07:54:09 +0200 Subject: ALSA: hda - Fix missing Mic Boost controls for VIA codecs Some VIA codecs like VT1708S have Mic boost amps in the mic pins but they aren't exposed in the capability bits. In the past driver code, we override the pin caps and create mic boost controls forcibly. While transition to the generic parser, we lost the mic boost controls although the pin caps are still overridden, because the generic parser code checks the widget caps, too. So this patch adds a new helper function to allow the override of the given widget capability bits, and makes VIA codecs driver to add the missing input-amp capability bit. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59861 Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 8 ++++++++ sound/pci/hda/patch_via.c | 2 ++ 2 files changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 29ed7d9b27e4..2e7493ef8ee0 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -562,6 +562,14 @@ static inline unsigned int get_wcaps_channels(u32 wcaps) return chans; } +static inline void snd_hda_override_wcaps(struct hda_codec *codec, + hda_nid_t nid, u32 val) +{ + if (nid >= codec->start_nid && + nid < codec->start_nid + codec->num_nodes) + codec->wcaps[nid - codec->start_nid] = val; +} + u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index cf31b664d2ed..dcebf3cb18de 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -905,6 +905,8 @@ static const struct hda_verb vt1708S_init_verbs[] = { static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, int offset, int num_steps, int step_size) { + snd_hda_override_wcaps(codec, pin, + get_wcaps(codec, pin) | AC_WCAP_IN_AMP); snd_hda_override_amp_caps(codec, pin, HDA_INPUT, (offset << AC_AMPCAP_OFFSET_SHIFT) | (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | -- cgit v1.2.1 From 1476f66f1f51aaf881842212ff73320a75014571 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:33:53 +0200 Subject: ASoC: tlv320aix3x: Use SOC_SINGLE_EXT() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 1514bf845e4b..e5b926883131 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -128,10 +128,8 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw_aic3x, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } + SOC_SINGLE_EXT(xname, reg, shift, mask, invert, \ + snd_soc_dapm_get_volsw, snd_soc_dapm_put_volsw_aic3x) /* * All input lines are connected when !0xf and disconnected with 0xf bit field, -- cgit v1.2.1 From a44b5177dc2f49b6dad68da3ba7db452892bd50e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:33:54 +0200 Subject: ASoC: wm8400: Use SOC_SINGLE_EXT_TLV() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index af6d227e67be..d2a092850283 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -143,13 +143,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, } #define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_get_volsw, .put = wm8400_outpga_put_volsw_vu, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT_TLV(xname, reg, shift, max, invert, \ + snd_soc_get_volsw, wm8400_outpga_put_volsw_vu, tlv_array) static const char *wm8400_digital_sidetone[] = -- cgit v1.2.1 From ea3583d04b35cacff6d5c3fabe5445c13bb89bfd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:33:55 +0200 Subject: ASoC: wm8903: Use SOC_SINGLE_EXT() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 9d88437cdcd1..fa24cedee687 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -403,10 +403,8 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, } #define SOC_DAPM_SINGLE_W(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = wm8903_class_w_put, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_dapm_get_volsw, wm8903_class_w_put) static int wm8903_deemph[] = { 0, 32000, 44100, 48000 }; -- cgit v1.2.1 From 5a68bae223636a581754f4f2a55f73a25cfe146c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:33:56 +0200 Subject: ASoC: wm8904: Use SOC_SINGLE_EXT() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3ff195c541db..4c9fb142cb2d 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -603,13 +603,8 @@ SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("High Pass Filter Mode", hpf_mode), - -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "ADC 128x OSR Switch", - .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, - .put = wm8904_adc_osr_put, - .private_value = SOC_SINGLE_VALUE(WM8904_ANALOGUE_ADC_0, 0, 1, 0), -}, +SOC_SINGLE_EXT("ADC 128x OSR Switch", WM8904_ANALOGUE_ADC_0, 0, 1, 0, + snd_soc_get_volsw, wm8904_adc_osr_put), }; static const char *drc_path_text[] = { -- cgit v1.2.1 From fc99adc3d82c3cbec9b12b2a638dbdd2a2e4ece1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:33:57 +0200 Subject: ASoC: wm8990: Use SOC_SINGLE_EXT_TLV() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 837978e16e9d..253c88bb7a4c 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -151,14 +151,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, } #define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ - tlv_array) {\ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + tlv_array) \ + SOC_SINGLE_EXT_TLV(xname, reg, shift, max, invert, \ + snd_soc_get_volsw, wm899x_outpga_put_volsw_vu, tlv_array) static const char *wm8990_digital_sidetone[] = -- cgit v1.2.1 From 9578121c80a59f2cdd93c8a48bc9549ee873b14c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:33:58 +0200 Subject: ASoC: wm8991: Use SOC_SINGLE_EXT_TLV() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h index 8a942efd18a5..07707d8d7e20 100644 --- a/sound/soc/codecs/wm8991.h +++ b/sound/soc/codecs/wm8991.h @@ -822,12 +822,7 @@ #define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT_TLV(xname, reg, shift, max, invert, \ + snd_soc_get_volsw, wm899x_outpga_put_volsw_vu, tlv_array) #endif /* _WM8991_H */ -- cgit v1.2.1 From 6e06509c583496fe24ffb498894bf2838b26492e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:33:59 +0200 Subject: ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29e95f93d482..9e13edd81292 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -289,10 +289,8 @@ static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ - .put = wm8994_put_drc_sw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, 1, 0) } + SOC_SINGLE_EXT(xname, reg, shift, 1, 0, \ + snd_soc_get_volsw, wm8994_put_drc_sw) static int wm8994_put_drc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1432,10 +1430,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, }; #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = wm8994_put_class_w, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_get_volsw, wm8994_put_class_w) static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.1 From d0a39cdcf264829fc07c2fcc5c6c8c98e01bc004 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:34:00 +0200 Subject: ASoC: wm8995: Use SOC_SINGLE_EXT() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.h | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.h b/sound/soc/codecs/wm8995.h index 5642121c4977..508ad27fe2bb 100644 --- a/sound/soc/codecs/wm8995.h +++ b/sound/soc/codecs/wm8995.h @@ -4237,11 +4237,8 @@ #define WM8995_SPK2_MUTE_SEQ1_WIDTH 8 /* SPK2_MUTE_SEQ1 - [7:0] */ #define WM8995_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = wm8995_put_class_w, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) \ -} + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_dapm_get_volsw, wm8995_put_class_w) struct wm8995_reg_access { u16 read; -- cgit v1.2.1 From 98809ae22b48e52d147695ab28999471835f5978 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:34:01 +0200 Subject: ASoC: wm_hubs: Use SOC_SINGLE_EXT() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f5d81b948759..2d9e099415a5 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -693,10 +693,8 @@ void wm_hubs_update_class_w(struct snd_soc_codec *codec) EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); #define WM_HUBS_SINGLE_W(xname, reg, shift, max, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = class_w_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ + snd_soc_dapm_get_volsw, class_w_put_volsw) static int class_w_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.1 From f9eeae9f04cafe2619be7ce17952dc502baeac39 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:34:02 +0200 Subject: ASoC: wm_adsp: Use SND_SOC_DAPM_PGA_E() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.h | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index fea514627526..36533eac420e 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -58,14 +58,12 @@ struct wm_adsp { }; #define WM_ADSP1(wname, num) \ - { .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ - .shift = num, .event = wm_adsp1_event, \ - .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } + SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ + wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) #define WM_ADSP2(wname, num) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ - .shift = num, .event = wm_adsp2_event, \ - .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } + SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ + wm_adsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; -- cgit v1.2.1 From f0ddb219c9f92d0c43582a000e1ddc0a4ee43222 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Jun 2013 19:34:03 +0200 Subject: ASoC: 88pm860x: Use SND_SOC_DAPM_PGA_E() instead of open-coding it Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 60159c07448d..e2bd3dd02c0e 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -120,10 +120,8 @@ * before DAC & PGA in DAPM power-off sequence. */ #define PM860X_DAPM_OUTPUT(wname, wevent) \ -{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ - .shift = 0, .invert = 0, .kcontrol_news = NULL, \ - .num_kcontrols = 0, .event = wevent, \ - .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, } + SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, 0, 0, NULL, 0, wevent, \ + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD) struct pm860x_det { struct snd_soc_jack *hp_jack; -- cgit v1.2.1 From 88d5760649d9024a2a68e649909f522ab42d891c Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 20 Jun 2013 10:23:54 +0200 Subject: ALSA: hda - Make Thinkpad X220-tablet use generic parser Like the X220, this quirk was added to support docking station, so enable the fixup instead. According to Jan, the generic parser works equal or better than the current parser. This was tested under a 3.9 kernel. Reported-by: Jan Alexander Steffens Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b314d3e6d7fa..de00ce166470 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2947,7 +2947,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), {} @@ -3318,6 +3317,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), -- cgit v1.2.1 From fefe228c5f13809f77e6b2873ffe8bfb006cadd4 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 21 Jun 2013 15:25:33 +0300 Subject: ALSA: vx_core: off by one in vx_read_status() This code is older than git, and I haven't tested it, but if size == SIZE_MAX_STATUS then we would write one space past the end of the rmh->Stat[] array. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index c39961c11401..83596891cde4 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -205,7 +205,7 @@ static int vx_read_status(struct vx_core *chip, struct vx_rmh *rmh) if (size < 1) return 0; - if (snd_BUG_ON(size > SIZE_MAX_STATUS)) + if (snd_BUG_ON(size >= SIZE_MAX_STATUS)) return -EINVAL; for (i = 1; i <= size; i++) { -- cgit v1.2.1 From 3dd446a7e5bb001ec681232ac63a558f5e0509ed Mon Sep 17 00:00:00 2001 From: Antonio Ospite Date: Fri, 21 Jun 2013 13:11:48 +0200 Subject: ALSA: snd-usb-caiaq: remove the unused snd_card_used variable The snd_card_used variable is only read but never written, remove it. Signed-off-by: Antonio Ospite Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 48b63ccc78c7..23cf55dbc54e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -57,7 +57,6 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int snd_card_used[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the caiaq sound device"); @@ -388,7 +387,7 @@ static int create_card(struct usb_device *usb_dev, struct snd_usb_caiaqdev *cdev; for (devnum = 0; devnum < SNDRV_CARDS; devnum++) - if (enable[devnum] && !snd_card_used[devnum]) + if (enable[devnum]) break; if (devnum >= SNDRV_CARDS) -- cgit v1.2.1 From fc76f8637650f422eb67385d5a584bf30befddaa Mon Sep 17 00:00:00 2001 From: Antonio Ospite Date: Fri, 21 Jun 2013 13:11:49 +0200 Subject: ALSA: snd-usb-caiaq: use vmalloc buffers For USB devices it's not necessary to allocate physically contiguous buffers. Signed-off-by: Antonio Ospite Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index c1916184e2e1..7103b0908d13 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -183,14 +183,15 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(sub, + params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); deactivate_substream(cdev, sub); - return snd_pcm_lib_free_pages(sub); + return snd_pcm_lib_free_vmalloc_buffer(sub); } /* this should probably go upstream */ @@ -345,7 +346,9 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .hw_free = snd_usb_caiaq_pcm_hw_free, .prepare = snd_usb_caiaq_pcm_prepare, .trigger = snd_usb_caiaq_pcm_trigger, - .pointer = snd_usb_caiaq_pcm_pointer + .pointer = snd_usb_caiaq_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, @@ -852,11 +855,6 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) snd_pcm_set_ops(cdev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); - snd_pcm_lib_preallocate_pages_for_all(cdev->pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); - cdev->data_cb_info = kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); -- cgit v1.2.1 From 4a9f9118619b20a7c10327667d6595b6bfa4beb1 Mon Sep 17 00:00:00 2001 From: Antonio Ospite Date: Fri, 21 Jun 2013 13:11:50 +0200 Subject: ALSA: snd-usb-6fire: use vmalloc buffers For USB devices it's not necessary to allocate physically contiguous buffers. Signed-off-by: Antonio Ospite Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 40dd50a80f55..c5b9cac37dc4 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -450,13 +450,13 @@ static int usb6fire_pcm_close(struct snd_pcm_substream *alsa_sub) static int usb6fire_pcm_hw_params(struct snd_pcm_substream *alsa_sub, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(alsa_sub, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(alsa_sub, + params_buffer_bytes(hw_params)); } static int usb6fire_pcm_hw_free(struct snd_pcm_substream *alsa_sub) { - return snd_pcm_lib_free_pages(alsa_sub); + return snd_pcm_lib_free_vmalloc_buffer(alsa_sub); } static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub) @@ -560,6 +560,8 @@ static struct snd_pcm_ops pcm_ops = { .prepare = usb6fire_pcm_prepare, .trigger = usb6fire_pcm_trigger, .pointer = usb6fire_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static void usb6fire_pcm_init_urb(struct pcm_urb *urb, @@ -622,10 +624,6 @@ int usb6fire_pcm_init(struct sfire_chip *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops); - ret = snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - MAX_BUFSIZE, MAX_BUFSIZE); if (ret) { kfree(rt); snd_printk(KERN_ERR PREFIX -- cgit v1.2.1 From 0af49ffe3c04ed5a9095ea1349d3e26a1e8b311a Mon Sep 17 00:00:00 2001 From: Antonio Ospite Date: Fri, 21 Jun 2013 13:11:51 +0200 Subject: ALSA: usb: uniform style used in MODULE_SUPPORTED_DEVICE() In sound/usb/card.c and sound/usb/misc/ua101.c there are no spaces between the vendor and the device names, use this style in the other drivers too. This also helps keeping consistency when new drivers copies from the ones already in the mainline tree. Signed-off-by: Antonio Ospite Signed-off-by: Takashi Iwai --- sound/usb/6fire/chip.c | 2 +- sound/usb/caiaq/device.c | 28 ++++++++++++++-------------- sound/usb/usx2y/usbusx2y.c | 2 +- 3 files changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index 4394ae796356..c39c77978468 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -30,7 +30,7 @@ MODULE_AUTHOR("Torsten Schenk "); MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); +MODULE_SUPPORTED_DEVICE("{{TerraTec,DMX 6Fire USB}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 23cf55dbc54e..1a61dd12fe38 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -39,20 +39,20 @@ MODULE_AUTHOR("Daniel Mack "); MODULE_DESCRIPTION("caiaq USB audio"); MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," - "{Native Instruments, RigKontrol3}," - "{Native Instruments, Kore Controller}," - "{Native Instruments, Kore Controller 2}," - "{Native Instruments, Audio Kontrol 1}," - "{Native Instruments, Audio 2 DJ}," - "{Native Instruments, Audio 4 DJ}," - "{Native Instruments, Audio 8 DJ}," - "{Native Instruments, Traktor Audio 2}," - "{Native Instruments, Session I/O}," - "{Native Instruments, GuitarRig mobile}," - "{Native Instruments, Traktor Kontrol X1}," - "{Native Instruments, Traktor Kontrol S4}," - "{Native Instruments, Maschine Controller}}"); +MODULE_SUPPORTED_DEVICE("{{Native Instruments,RigKontrol2}," + "{Native Instruments,RigKontrol3}," + "{Native Instruments,Kore Controller}," + "{Native Instruments,Kore Controller 2}," + "{Native Instruments,Audio Kontrol 1}," + "{Native Instruments,Audio 2 DJ}," + "{Native Instruments,Audio 4 DJ}," + "{Native Instruments,Audio 8 DJ}," + "{Native Instruments,Traktor Audio 2}," + "{Native Instruments,Session I/O}," + "{Native Instruments,GuitarRig mobile}," + "{Native Instruments,Traktor Kontrol X1}," + "{Native Instruments,Traktor Kontrol S4}," + "{Native Instruments,Maschine Controller}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 9af7c1f17413..1f9bbd55553f 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -150,7 +150,7 @@ MODULE_AUTHOR("Karsten Wiese "); MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.8.7.2"); MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604), "NAME_ALLCAPS"(0x8001)(0x8005)(0x8007) }}"); +MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604),"NAME_ALLCAPS"(0x8001)(0x8005)(0x8007)}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ -- cgit v1.2.1 From 26889e51d39cc5cb36685f5a48a612108598f89c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 21 Jun 2013 18:02:01 +0200 Subject: ASoC: omap: Fix the leftover CONFIG_SND_SOC_HDMI_CODEC Replace the leftover CONFIG_SND_SOC_OMAP_HDMI_CODEC in sound/soc/omap/Kconfig with CONFIG_SND_SOC_HDMI_CODEC, which was forgotten in the commit bf7c6e6c. Signed-off-by: Takashi Iwai --- sound/soc/omap/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 60259f2f3f2c..9f5d55e6b17a 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -103,7 +103,7 @@ config SND_OMAP_SOC_OMAP_HDMI tristate "SoC Audio support for Texas Instruments OMAP HDMI" depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS select SND_OMAP_SOC_HDMI - select SND_SOC_OMAP_HDMI_CODEC + select SND_SOC_HDMI_CODEC select OMAP4_DSS_HDMI_AUDIO help Say Y if you want to add support for SoC HDMI audio on Texas Instruments -- cgit v1.2.1 From 04c9548c78f7bb00c438cb8005bf4ec25b5720fe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 21 Jun 2013 18:09:49 +0200 Subject: ASoC: samsung: Fix a typo of CONFIG_SND_SOC_BT_SCO ... instead of CONFIG_SND_SOC_SCO. Introduced in the commit 200ceb96. Signed-off-by: Takashi Iwai --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index ae0ea87b7d7b..9855dfc3e3ec 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,7 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_SCO + select SND_SOC_BT_SCO help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. -- cgit v1.2.1 From a91c3fb2f84204dcf024ca6a032f12cdb84f2196 Mon Sep 17 00:00:00 2001 From: Antonio Ospite Date: Sat, 22 Jun 2013 00:14:46 +0200 Subject: Add M2Tech hiFace USB-SPDIF driver Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices. M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output Interface, see http://www.m2tech.biz/hiface.html The supported products are: * M2Tech Young * M2Tech hiFace * M2Tech North Star * M2Tech W4S Young * M2Tech Corrson * M2Tech AUDIA * M2Tech SL Audio * M2Tech Empirical * M2Tech Rockna * M2Tech Pathos * M2Tech Metronome * M2Tech CAD * M2Tech Audio Esclusive * M2Tech Rotel * M2Tech Eeaudio * The Chord Company CHORD * AVA Group A/S Vitus Signed-off-by: Antonio Ospite Signed-off-by: Takashi Iwai --- sound/usb/Kconfig | 31 +++ sound/usb/Makefile | 2 +- sound/usb/hiface/Makefile | 2 + sound/usb/hiface/chip.c | 297 ++++++++++++++++++++++ sound/usb/hiface/chip.h | 30 +++ sound/usb/hiface/pcm.c | 621 ++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/hiface/pcm.h | 24 ++ 7 files changed, 1006 insertions(+), 1 deletion(-) create mode 100644 sound/usb/hiface/Makefile create mode 100644 sound/usb/hiface/chip.c create mode 100644 sound/usb/hiface/chip.h create mode 100644 sound/usb/hiface/pcm.c create mode 100644 sound/usb/hiface/pcm.h (limited to 'sound') diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd737265..de9408b83f75 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -115,5 +115,36 @@ config SND_USB_6FIRE and further help can be found at http://sixfireusb.sourceforge.net +config SND_USB_HIFACE + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help + Select this option to include support for M2Tech hiFace USB-SPDIF + interface. + + This driver supports the original M2Tech hiFace and some other + compatible devices. The supported products are: + + * M2Tech Young + * M2Tech hiFace + * M2Tech North Star + * M2Tech W4S Young + * M2Tech Corrson + * M2Tech AUDIA + * M2Tech SL Audio + * M2Tech Empirical + * M2Tech Rockna + * M2Tech Pathos + * M2Tech Metronome + * M2Tech CAD + * M2Tech Audio Esclusive + * M2Tech Rotel + * M2Tech Eeaudio + * The Chord Company CHORD + * AVA Group A/S Vitus + + To compile this driver as a module, choose M here: the module + will be called snd-usb-hiface. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index ac256dc4c6be..abe668f660d1 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile new file mode 100644 index 000000000000..463b136d1d89 --- /dev/null +++ b/sound/usb/hiface/Makefile @@ -0,0 +1,2 @@ +snd-usb-hiface-objs := chip.o pcm.o +obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c new file mode 100644 index 000000000000..b0dcb3924ce5 --- /dev/null +++ b/sound/usb/hiface/chip.c @@ -0,0 +1,297 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi + * Antonio Ospite + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include +#include +#include + +#include "chip.h" +#include "pcm.h" + +MODULE_AUTHOR("Michael Trimarchi "); +MODULE_AUTHOR("Antonio Ospite "); +MODULE_DESCRIPTION("M2Tech hiFace USB-SPDIF audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{M2Tech,Young}," + "{M2Tech,hiFace}," + "{M2Tech,North Star}," + "{M2Tech,W4S Young}," + "{M2Tech,Corrson}," + "{M2Tech,AUDIA}," + "{M2Tech,SL Audio}," + "{M2Tech,Empirical}," + "{M2Tech,Rockna}," + "{M2Tech,Pathos}," + "{M2Tech,Metronome}," + "{M2Tech,CAD}," + "{M2Tech,Audio Esclusive}," + "{M2Tech,Rotel}," + "{M2Tech,Eeaudio}," + "{The Chord Company,CHORD}," + "{AVA Group A/S,Vitus}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +#define DRIVER_NAME "snd-usb-hiface" +#define CARD_NAME "hiFace" + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +static DEFINE_MUTEX(register_mutex); + +struct hiface_vendor_quirk { + const char *device_name; + u8 extra_freq; +}; + +static int hiface_chip_create(struct usb_device *device, int idx, + const struct hiface_vendor_quirk *quirk, + struct hiface_chip **rchip) +{ + struct snd_card *card = NULL; + struct hiface_chip *chip; + int ret; + int len; + + *rchip = NULL; + + /* if we are here, card can be registered in alsa. */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, sizeof(*chip), &card); + if (ret < 0) { + dev_err(&device->dev, "cannot create alsa card.\n"); + return ret; + } + + strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver)); + + if (quirk && quirk->device_name) + strlcpy(card->shortname, quirk->device_name, sizeof(card->shortname)); + else + strlcpy(card->shortname, "M2Tech generic audio", sizeof(card->shortname)); + + strlcat(card->longname, card->shortname, sizeof(card->longname)); + len = strlcat(card->longname, " at ", sizeof(card->longname)); + if (len < sizeof(card->longname)) + usb_make_path(device, card->longname + len, + sizeof(card->longname) - len); + + chip = card->private_data; + chip->dev = device; + chip->card = card; + + *rchip = chip; + return 0; +} + +static int hiface_chip_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct hiface_vendor_quirk *quirk = (struct hiface_vendor_quirk *)usb_id->driver_info; + int ret; + int i; + struct hiface_chip *chip; + struct usb_device *device = interface_to_usbdev(intf); + + ret = usb_set_interface(device, 0, 0); + if (ret != 0) { + dev_err(&device->dev, "can't set first interface for " CARD_NAME " device.\n"); + return -EIO; + } + + /* check whether the card is already registered */ + chip = NULL; + mutex_lock(®ister_mutex); + + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i]) + break; + + if (i >= SNDRV_CARDS) { + dev_err(&device->dev, "no available " CARD_NAME " audio device\n"); + ret = -ENODEV; + goto err; + } + + ret = hiface_chip_create(device, i, quirk, &chip); + if (ret < 0) + goto err; + + snd_card_set_dev(chip->card, &intf->dev); + + ret = hiface_pcm_init(chip, quirk ? quirk->extra_freq : 0); + if (ret < 0) + goto err_chip_destroy; + + ret = snd_card_register(chip->card); + if (ret < 0) { + dev_err(&device->dev, "cannot register " CARD_NAME " card\n"); + goto err_chip_destroy; + } + + mutex_unlock(®ister_mutex); + + usb_set_intfdata(intf, chip); + return 0; + +err_chip_destroy: + snd_card_free(chip->card); +err: + mutex_unlock(®ister_mutex); + return ret; +} + +static void hiface_chip_disconnect(struct usb_interface *intf) +{ + struct hiface_chip *chip; + struct snd_card *card; + + chip = usb_get_intfdata(intf); + if (!chip) + return; + + card = chip->card; + + /* Make sure that the userspace cannot create new request */ + snd_card_disconnect(card); + + hiface_pcm_abort(chip); + snd_card_free_when_closed(card); +} + +static const struct usb_device_id device_table[] = { + { + USB_DEVICE(0x04b4, 0x0384), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Young", + .extra_freq = 1, + } + }, + { + USB_DEVICE(0x04b4, 0x930b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "hiFace", + } + }, + { + USB_DEVICE(0x04b4, 0x931b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "North Star", + } + }, + { + USB_DEVICE(0x04b4, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "W4S Young", + } + }, + { + USB_DEVICE(0x04b4, 0x931d), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Corrson", + } + }, + { + USB_DEVICE(0x04b4, 0x931e), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "AUDIA", + } + }, + { + USB_DEVICE(0x04b4, 0x931f), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "SL Audio", + } + }, + { + USB_DEVICE(0x04b4, 0x9320), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Empirical", + } + }, + { + USB_DEVICE(0x04b4, 0x9321), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rockna", + } + }, + { + USB_DEVICE(0x249c, 0x9001), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Pathos", + } + }, + { + USB_DEVICE(0x249c, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Metronome", + } + }, + { + USB_DEVICE(0x249c, 0x9006), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CAD", + } + }, + { + USB_DEVICE(0x249c, 0x9008), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Audio Esclusive", + } + }, + { + USB_DEVICE(0x249c, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rotel", + } + }, + { + USB_DEVICE(0x249c, 0x932c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Eeaudio", + } + }, + { + USB_DEVICE(0x245f, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CHORD", + } + }, + { + USB_DEVICE(0x25c6, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Vitus", + } + }, + {} +}; + +MODULE_DEVICE_TABLE(usb, device_table); + +static struct usb_driver hiface_usb_driver = { + .name = DRIVER_NAME, + .probe = hiface_chip_probe, + .disconnect = hiface_chip_disconnect, + .id_table = device_table, +}; + +module_usb_driver(hiface_usb_driver); diff --git a/sound/usb/hiface/chip.h b/sound/usb/hiface/chip.h new file mode 100644 index 000000000000..189a1371b7c4 --- /dev/null +++ b/sound/usb/hiface/chip.h @@ -0,0 +1,30 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi + * Antonio Ospite + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_CHIP_H +#define HIFACE_CHIP_H + +#include +#include + +struct pcm_runtime; + +struct hiface_chip { + struct usb_device *dev; + struct snd_card *card; + struct pcm_runtime *pcm; +}; +#endif /* HIFACE_CHIP_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c new file mode 100644 index 000000000000..6430ed2a9f65 --- /dev/null +++ b/sound/usb/hiface/pcm.c @@ -0,0 +1,621 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi + * Antonio Ospite + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include +#include + +#include "pcm.h" +#include "chip.h" + +#define OUT_EP 0x2 +#define PCM_N_URBS 8 +#define PCM_PACKET_SIZE 4096 +#define PCM_BUFFER_SIZE (2 * PCM_N_URBS * PCM_PACKET_SIZE) + +struct pcm_urb { + struct hiface_chip *chip; + + struct urb instance; + struct usb_anchor submitted; + u8 *buffer; +}; + +struct pcm_substream { + spinlock_t lock; + struct snd_pcm_substream *instance; + + bool active; + snd_pcm_uframes_t dma_off; /* current position in alsa dma_area */ + snd_pcm_uframes_t period_off; /* current position in current period */ +}; + +enum { /* pcm streaming states */ + STREAM_DISABLED, /* no pcm streaming */ + STREAM_STARTING, /* pcm streaming requested, waiting to become ready */ + STREAM_RUNNING, /* pcm streaming running */ + STREAM_STOPPING +}; + +struct pcm_runtime { + struct hiface_chip *chip; + struct snd_pcm *instance; + + struct pcm_substream playback; + bool panic; /* if set driver won't do anymore pcm on device */ + + struct pcm_urb out_urbs[PCM_N_URBS]; + + struct mutex stream_mutex; + u8 stream_state; /* one of STREAM_XXX */ + u8 extra_freq; + wait_queue_head_t stream_wait_queue; + bool stream_wait_cond; +}; + +static const unsigned int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000 }; +static const struct snd_pcm_hw_constraint_list constraints_extra_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const struct snd_pcm_hardware pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, + + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + + .rate_min = 44100, + .rate_max = 192000, /* changes in hiface_pcm_open to support extra rates */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = PCM_BUFFER_SIZE, + .period_bytes_min = PCM_PACKET_SIZE, + .period_bytes_max = PCM_BUFFER_SIZE, + .periods_min = 2, + .periods_max = 1024 +}; + +/* message values used to change the sample rate */ +#define HIFACE_SET_RATE_REQUEST 0xb0 + +#define HIFACE_RATE_44100 0x43 +#define HIFACE_RATE_48000 0x4b +#define HIFACE_RATE_88200 0x42 +#define HIFACE_RATE_96000 0x4a +#define HIFACE_RATE_176400 0x40 +#define HIFACE_RATE_192000 0x48 +#define HIFACE_RATE_352000 0x58 +#define HIFACE_RATE_384000 0x68 + +static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) +{ + struct usb_device *device = rt->chip->dev; + u16 rate_value; + int ret; + + /* We are already sure that the rate is supported here thanks to + * ALSA constraints + */ + switch (rate) { + case 44100: + rate_value = HIFACE_RATE_44100; + break; + case 48000: + rate_value = HIFACE_RATE_48000; + break; + case 88200: + rate_value = HIFACE_RATE_88200; + break; + case 96000: + rate_value = HIFACE_RATE_96000; + break; + case 176400: + rate_value = HIFACE_RATE_176400; + break; + case 192000: + rate_value = HIFACE_RATE_192000; + break; + case 352000: + rate_value = HIFACE_RATE_352000; + break; + case 384000: + rate_value = HIFACE_RATE_384000; + break; + default: + dev_err(&device->dev, "Unsupported rate %d\n", rate); + return -EINVAL; + } + + /* + * USBIO: Vendor 0xb0(wValue=0x0043, wIndex=0x0000) + * 43 b0 43 00 00 00 00 00 + * USBIO: Vendor 0xb0(wValue=0x004b, wIndex=0x0000) + * 43 b0 4b 00 00 00 00 00 + * This control message doesn't have any ack from the + * other side + */ + ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), + HIFACE_SET_RATE_REQUEST, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + rate_value, 0, NULL, 0, 100); + if (ret < 0) { + dev_err(&device->dev, "Error setting samplerate %d.\n", rate); + return ret; + } + + return 0; +} + +static struct pcm_substream *hiface_pcm_get_substream(struct snd_pcm_substream + *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct device *device = &rt->chip->dev->dev; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rt->playback; + + dev_err(device, "Error getting pcm substream slot.\n"); + return NULL; +} + +/* call with stream_mutex locked */ +static void hiface_pcm_stream_stop(struct pcm_runtime *rt) +{ + int i, time; + + if (rt->stream_state != STREAM_DISABLED) { + rt->stream_state = STREAM_STOPPING; + + for (i = 0; i < PCM_N_URBS; i++) { + time = usb_wait_anchor_empty_timeout( + &rt->out_urbs[i].submitted, 100); + if (!time) + usb_kill_anchored_urbs( + &rt->out_urbs[i].submitted); + usb_kill_urb(&rt->out_urbs[i].instance); + } + + rt->stream_state = STREAM_DISABLED; + } +} + +/* call with stream_mutex locked */ +static int hiface_pcm_stream_start(struct pcm_runtime *rt) +{ + int ret = 0; + int i; + + if (rt->stream_state == STREAM_DISABLED) { + + /* reset panic state when starting a new stream */ + rt->panic = false; + + /* submit our out urbs zero init */ + rt->stream_state = STREAM_STARTING; + for (i = 0; i < PCM_N_URBS; i++) { + memset(rt->out_urbs[i].buffer, 0, PCM_PACKET_SIZE); + usb_anchor_urb(&rt->out_urbs[i].instance, + &rt->out_urbs[i].submitted); + ret = usb_submit_urb(&rt->out_urbs[i].instance, + GFP_ATOMIC); + if (ret) { + hiface_pcm_stream_stop(rt); + return ret; + } + } + + /* wait for first out urb to return (sent in in urb handler) */ + wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, + HZ); + if (rt->stream_wait_cond) { + struct device *device = &rt->chip->dev->dev; + dev_dbg(device, "%s: Stream is running wakeup event\n", + __func__); + rt->stream_state = STREAM_RUNNING; + } else { + hiface_pcm_stream_stop(rt); + return -EIO; + } + } + return ret; +} + +/* The hardware wants word-swapped 32-bit values */ +static void memcpy_swahw32(u8 *dest, u8 *src, unsigned int n) +{ + unsigned int i; + + for (i = 0; i < n / 4; i++) + ((u32 *)dest)[i] = swahw32(((u32 *)src)[i]); +} + +/* call with substream locked */ +/* returns true if a period elapsed */ +static bool hiface_pcm_playback(struct pcm_substream *sub, struct pcm_urb *urb) +{ + struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; + struct device *device = &urb->chip->dev->dev; + u8 *source; + unsigned int pcm_buffer_size; + + WARN_ON(alsa_rt->format != SNDRV_PCM_FORMAT_S32_LE); + + pcm_buffer_size = snd_pcm_lib_buffer_bytes(sub->instance); + + if (sub->dma_off + PCM_PACKET_SIZE <= pcm_buffer_size) { + dev_dbg(device, "%s: (1) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, PCM_PACKET_SIZE); + } else { + /* wrap around at end of ring buffer */ + unsigned int len; + + dev_dbg(device, "%s: (2) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + len = pcm_buffer_size - sub->dma_off; + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, len); + + source = alsa_rt->dma_area; + memcpy_swahw32(urb->buffer + len, source, + PCM_PACKET_SIZE - len); + } + sub->dma_off += PCM_PACKET_SIZE; + if (sub->dma_off >= pcm_buffer_size) + sub->dma_off -= pcm_buffer_size; + + sub->period_off += PCM_PACKET_SIZE; + if (sub->period_off >= alsa_rt->period_size) { + sub->period_off %= alsa_rt->period_size; + return true; + } + return false; +} + +static void hiface_pcm_out_urb_handler(struct urb *usb_urb) +{ + struct pcm_urb *out_urb = usb_urb->context; + struct pcm_runtime *rt = out_urb->chip->pcm; + struct pcm_substream *sub; + bool do_period_elapsed = false; + unsigned long flags; + int ret; + + if (rt->panic || rt->stream_state == STREAM_STOPPING) + return; + + if (unlikely(usb_urb->status == -ENOENT || /* unlinked */ + usb_urb->status == -ENODEV || /* device removed */ + usb_urb->status == -ECONNRESET || /* unlinked */ + usb_urb->status == -ESHUTDOWN)) { /* device disabled */ + goto out_fail; + } + + if (rt->stream_state == STREAM_STARTING) { + rt->stream_wait_cond = true; + wake_up(&rt->stream_wait_queue); + } + + /* now send our playback data (if a free out urb was found) */ + sub = &rt->playback; + spin_lock_irqsave(&sub->lock, flags); + if (sub->active) + do_period_elapsed = hiface_pcm_playback(sub, out_urb); + else + memset(out_urb->buffer, 0, PCM_PACKET_SIZE); + + spin_unlock_irqrestore(&sub->lock, flags); + + if (do_period_elapsed) + snd_pcm_period_elapsed(sub->instance); + + ret = usb_submit_urb(&out_urb->instance, GFP_ATOMIC); + if (ret < 0) + goto out_fail; + + return; + +out_fail: + rt->panic = true; +} + +static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = NULL; + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + if (rt->panic) + return -EPIPE; + + mutex_lock(&rt->stream_mutex); + alsa_rt->hw = pcm_hw; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + sub = &rt->playback; + + if (!sub) { + struct device *device = &rt->chip->dev->dev; + mutex_unlock(&rt->stream_mutex); + dev_err(device, "Invalid stream type\n"); + return -EINVAL; + } + + if (rt->extra_freq) { + alsa_rt->hw.rates |= SNDRV_PCM_RATE_KNOT; + alsa_rt->hw.rate_max = 384000; + + /* explicit constraints needed as we added SNDRV_PCM_RATE_KNOT */ + ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_extra_rates); + if (ret < 0) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + + sub->instance = alsa_sub; + sub->active = false; + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + unsigned long flags; + + if (rt->panic) + return 0; + + mutex_lock(&rt->stream_mutex); + if (sub) { + hiface_pcm_stream_stop(rt); + + /* deactivate substream */ + spin_lock_irqsave(&sub->lock, flags); + sub->instance = NULL; + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_alloc_vmalloc_buffer(alsa_sub, + params_buffer_bytes(hw_params)); +} + +static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) +{ + return snd_pcm_lib_free_vmalloc_buffer(alsa_sub); +} + +static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + mutex_lock(&rt->stream_mutex); + + sub->dma_off = 0; + sub->period_off = 0; + + if (rt->stream_state == STREAM_DISABLED) { + + ret = hiface_pcm_set_rate(rt, alsa_rt->rate); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + ret = hiface_pcm_stream_start(rt); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_trigger(struct snd_pcm_substream *alsa_sub, int cmd) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irq(&sub->lock); + sub->active = true; + spin_unlock_irq(&sub->lock); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irq(&sub->lock); + sub->active = false; + spin_unlock_irq(&sub->lock); + return 0; + + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + snd_pcm_uframes_t dma_offset; + + if (rt->panic || !sub) + return SNDRV_PCM_STATE_XRUN; + + spin_lock_irqsave(&sub->lock, flags); + dma_offset = sub->dma_off; + spin_unlock_irqrestore(&sub->lock, flags); + return bytes_to_frames(alsa_sub->runtime, dma_offset); +} + +static struct snd_pcm_ops pcm_ops = { + .open = hiface_pcm_open, + .close = hiface_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hiface_pcm_hw_params, + .hw_free = hiface_pcm_hw_free, + .prepare = hiface_pcm_prepare, + .trigger = hiface_pcm_trigger, + .pointer = hiface_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +static int hiface_pcm_init_urb(struct pcm_urb *urb, + struct hiface_chip *chip, + unsigned int ep, + void (*handler)(struct urb *)) +{ + urb->chip = chip; + usb_init_urb(&urb->instance); + + urb->buffer = kzalloc(PCM_PACKET_SIZE, GFP_KERNEL); + if (!urb->buffer) + return -ENOMEM; + + usb_fill_bulk_urb(&urb->instance, chip->dev, + usb_sndbulkpipe(chip->dev, ep), (void *)urb->buffer, + PCM_PACKET_SIZE, handler, urb); + init_usb_anchor(&urb->submitted); + + return 0; +} + +void hiface_pcm_abort(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + + if (rt) { + rt->panic = true; + + mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + mutex_unlock(&rt->stream_mutex); + } +} + +static void hiface_pcm_destroy(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + int i; + + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + + kfree(chip->pcm); + chip->pcm = NULL; +} + +static void hiface_pcm_free(struct snd_pcm *pcm) +{ + struct pcm_runtime *rt = pcm->private_data; + + if (rt) + hiface_pcm_destroy(rt->chip); +} + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) +{ + int i; + int ret; + struct snd_pcm *pcm; + struct pcm_runtime *rt; + + rt = kzalloc(sizeof(*rt), GFP_KERNEL); + if (!rt) + return -ENOMEM; + + rt->chip = chip; + rt->stream_state = STREAM_DISABLED; + if (extra_freq) + rt->extra_freq = 1; + + init_waitqueue_head(&rt->stream_wait_queue); + mutex_init(&rt->stream_mutex); + spin_lock_init(&rt->playback.lock); + + for (i = 0; i < PCM_N_URBS; i++) + hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, + hiface_pcm_out_urb_handler); + + ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); + if (ret < 0) { + kfree(rt); + dev_err(&chip->dev->dev, "Cannot create pcm instance\n"); + return ret; + } + + pcm->private_data = rt; + pcm->private_free = hiface_pcm_free; + + strlcpy(pcm->name, "USB-SPDIF Audio", sizeof(pcm->name)); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + + rt->instance = pcm; + + chip->pcm = rt; + return 0; +} diff --git a/sound/usb/hiface/pcm.h b/sound/usb/hiface/pcm.h new file mode 100644 index 000000000000..77edd7c12e19 --- /dev/null +++ b/sound/usb/hiface/pcm.h @@ -0,0 +1,24 @@ +/* + * Linux driver for M2Tech hiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi + * Antonio Ospite + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_PCM_H +#define HIFACE_PCM_H + +struct hiface_chip; + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq); +void hiface_pcm_abort(struct hiface_chip *chip); +#endif /* HIFACE_PCM_H */ -- cgit v1.2.1 From 3af3f356e16041c3353210214da601782e2cd8b4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 24 Jun 2013 10:18:54 -0400 Subject: ALSA: hda - reset hda link during system/runtime suspend If all the codecs report ClkStopOK (OK to stop bus clock) after being put to D3, this patch will reset the HDA link before the controller is put to D3. So the link will be in reset during system or runtime suspend, the bus clock stops and the codecs are in D3(ClkStop) state. This may help to reduce power consumption by dozens of mW on some peripheral hda codecs. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f089fa0aa03d..9f110c7ba092 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1120,6 +1120,20 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, struct snd_dma_buffer *dmab); #endif +/* enter link reset */ +static void azx_reset_link(struct azx *chip) +{ + unsigned long timeout; + + /* reset controller */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); + + timeout = jiffies + msecs_to_jiffies(100); + while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); +} + /* reset codec link */ static int azx_reset(struct azx *chip, int full_reset) { @@ -2894,6 +2908,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); + azx_reset_link(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2946,6 +2961,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; azx_stop_chip(chip); + azx_reset_link(chip); azx_clear_irq_pending(chip); return 0; } -- cgit v1.2.1 From e1a4dca6711c68b6fcc4a236b3475f25dbf227ae Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 24 Jun 2013 09:10:18 -0400 Subject: ALSA: hda - Remove unused variable Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 49ef8f8eb5e9..8428763de153 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1887,7 +1887,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -- cgit v1.2.1 From 1ca2f2ec9e74e9d6e398e09b6468b4462c6d6b6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 24 Jun 2013 15:51:54 +0200 Subject: ALSA: vmaster: Add snd_ctl_sync_vmaster() helper function Introduce a new helper function, snd_ctl_sync_vmaster(), which updates the slave put callbacks forcibly as well as calling the hook. This will be used in the upcoming patch in HD-audio codec driver for toggling the mute in vmaster slaves. Along with the new function, the old snd_ctl_sync_vmaster_hook() is replaced as a macro calling with the argument hook_only=true. Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 65 ++++++++++++++++++++++++++++++++++++---------------- 1 file changed, 45 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 02f90b4f8b86..5df8dc25ad80 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -310,20 +310,10 @@ static int master_get(struct snd_kcontrol *kcontrol, return 0; } -static int master_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int sync_slaves(struct link_master *master, int old_val, int new_val) { - struct link_master *master = snd_kcontrol_chip(kcontrol); struct link_slave *slave; struct snd_ctl_elem_value *uval; - int err, old_val; - - err = master_init(master); - if (err < 0) - return err; - old_val = master->val; - if (ucontrol->value.integer.value[0] == old_val) - return 0; uval = kmalloc(sizeof(*uval), GFP_KERNEL); if (!uval) @@ -332,11 +322,33 @@ static int master_put(struct snd_kcontrol *kcontrol, master->val = old_val; uval->id = slave->slave.id; slave_get_val(slave, uval); - master->val = ucontrol->value.integer.value[0]; + master->val = new_val; slave_put_val(slave, uval); } kfree(uval); - if (master->hook && !err) + return 0; +} + +static int master_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + int err, new_val, old_val; + bool first_init; + + err = master_init(master); + if (err < 0) + return err; + first_init = err; + old_val = master->val; + new_val = ucontrol->value.integer.value[0]; + if (new_val == old_val) + return 0; + + err = sync_slaves(master, old_val, new_val); + if (err < 0) + return err; + if (master->hook && first_init) master->hook(master->hook_private_data, master->val); return 1; } @@ -442,20 +454,33 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook); /** - * snd_ctl_sync_vmaster_hook - Sync the vmaster hook + * snd_ctl_sync_vmaster - Sync the vmaster slaves and hook * @kcontrol: vmaster kctl element + * @hook_only: sync only the hook * - * Call the hook function to synchronize with the current value of the given - * vmaster element. NOP when NULL is passed to @kcontrol or the hook doesn't - * exist. + * Forcibly call the put callback of each slave and call the hook function + * to synchronize with the current value of the given vmaster element. + * NOP when NULL is passed to @kcontrol. */ -void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kcontrol) +void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only) { struct link_master *master; + bool first_init = false; + if (!kcontrol) return; master = snd_kcontrol_chip(kcontrol); - if (master->hook) + if (!hook_only) { + int err = master_init(master); + if (err < 0) + return; + first_init = err; + err = sync_slaves(master, master->val, master->val); + if (err < 0) + return; + } + + if (master->hook && !first_init) master->hook(master->hook_private_data, master->val); } -EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster_hook); +EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster); -- cgit v1.2.1 From f60596d61fc238befd169ea394ba6a458fafd774 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 24 Jun 2013 15:42:03 +0200 Subject: ASoC: twl6040: Drop using devm_request_threaded_irq() We need to free the irq at twl6040_remove() which is called when the machine driver has been removed (the card has been removed). If we fail to do that, next time when the machine driver is loaded the codec's probe will fail since the irq has been already requested. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9b9a6e587610..c2f2fdbfef96 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1143,7 +1143,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) mutex_init(&priv->mutex); - ret = devm_request_threaded_irq(codec->dev, priv->plug_irq, NULL, + ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, IRQF_NO_SUSPEND, "twl6040_irq_plug", codec); if (ret) { @@ -1159,6 +1159,9 @@ static int twl6040_probe(struct snd_soc_codec *codec) static int twl6040_remove(struct snd_soc_codec *codec) { + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + + free_irq(priv->plug_irq, codec); twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; -- cgit v1.2.1 From 68897497aa2e1eb9c3a6b55f8212cd1edc22acd5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 24 Jun 2013 15:42:05 +0200 Subject: ASoC: twl6040: Assign id for each DAI Later we can identify the DAIs by this ID number. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c2f2fdbfef96..9ea3dbccc0b3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -38,6 +38,14 @@ #include "twl6040.h" +enum twl6040_dai_id { + TWL6040_DAI_LEGACY = 0, + TWL6040_DAI_UL, + TWL6040_DAI_DL1, + TWL6040_DAI_DL2, + TWL6040_DAI_VIB, +}; + #define TWL6040_RATES SNDRV_PCM_RATE_8000_96000 #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) @@ -1036,6 +1044,7 @@ static const struct snd_soc_dai_ops twl6040_dai_ops = { static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-legacy", + .id = TWL6040_DAI_LEGACY, .playback = { .stream_name = "Legacy Playback", .channels_min = 1, @@ -1054,6 +1063,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-ul", + .id = TWL6040_DAI_UL, .capture = { .stream_name = "Capture", .channels_min = 1, @@ -1065,6 +1075,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-dl1", + .id = TWL6040_DAI_DL1, .playback = { .stream_name = "Headset Playback", .channels_min = 1, @@ -1076,6 +1087,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-dl2", + .id = TWL6040_DAI_DL2, .playback = { .stream_name = "Handsfree Playback", .channels_min = 1, @@ -1087,6 +1099,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, { .name = "twl6040-vib", + .id = TWL6040_DAI_VIB, .playback = { .stream_name = "Vibra Playback", .channels_min = 1, -- cgit v1.2.1 From 98c5fb1f875732e49ce223ba920204ec57f51511 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 24 Jun 2013 15:42:06 +0200 Subject: ASoC: twl6040: Add digital mute support To reduce pop noise during playback stream start and stop the codec needs to have the digital_mute callback implemented. The codec need to be muted before the CPU dai has been stopped (McPDM). Stopping the McPDM will generate a pop on the codec since no signal on the PDM bus means full negative amplitude. By managing the mute/unmute state of the outputs we can decrease the amount of pop noise when playback starts or stops. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 91 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 90 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9ea3dbccc0b3..44621ddc332d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -75,6 +75,8 @@ struct twl6040_data { int pll_power_mode; int hs_power_mode; int hs_power_mode_locked; + bool dl1_unmuted; + bool dl2_unmuted; unsigned int clk_in; unsigned int sysclk; struct twl6040_jack_data hs_jack; @@ -228,6 +230,25 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, return value; } +static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + + switch (reg) { + case TWL6040_REG_HSLCTL: + case TWL6040_REG_HSRCTL: + case TWL6040_REG_EARCTL: + /* DL1 path */ + return priv->dl1_unmuted; + case TWL6040_REG_HFLCTL: + case TWL6040_REG_HFRCTL: + return priv->dl2_unmuted; + default: + return 1; + }; +} + /* * write to the twl6040 register space */ @@ -240,7 +261,8 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - if (likely(reg < TWL6040_REG_SW_SHADOW)) + if (likely(reg < TWL6040_REG_SW_SHADOW) && + twl6040_is_path_unmuted(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else return 0; @@ -1034,11 +1056,78 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } +static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id id, + int mute) +{ + struct twl6040 *twl6040 = codec->control_data; + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + int hslctl, hsrctl, earctl; + int hflctl, hfrctl; + + switch (id) { + case TWL6040_DAI_DL1: + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + earctl = twl6040_read_reg_cache(codec, TWL6040_REG_EARCTL); + + if (mute) { + /* Power down drivers and DACs */ + earctl &= ~0x01; + hslctl &= ~(TWL6040_HSDRVENA | TWL6040_HSDACENA); + hsrctl &= ~(TWL6040_HSDRVENA | TWL6040_HSDACENA); + + } + + twl6040_reg_write(twl6040, TWL6040_REG_EARCTL, earctl); + twl6040_reg_write(twl6040, TWL6040_REG_HSLCTL, hslctl); + twl6040_reg_write(twl6040, TWL6040_REG_HSRCTL, hsrctl); + priv->dl1_unmuted = !mute; + break; + case TWL6040_DAI_DL2: + hflctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFLCTL); + hfrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFRCTL); + + if (mute) { + /* Power down drivers and DACs */ + hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | + TWL6040_HFDRVENA); + hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | + TWL6040_HFDRVENA); + } + + twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl); + twl6040_reg_write(twl6040, TWL6040_REG_HFRCTL, hfrctl); + priv->dl2_unmuted = !mute; + break; + default: + break; + }; +} + +static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) +{ + switch (dai->id) { + case TWL6040_DAI_LEGACY: + twl6040_mute_path(dai->codec, TWL6040_DAI_DL1, mute); + twl6040_mute_path(dai->codec, TWL6040_DAI_DL2, mute); + break; + case TWL6040_DAI_DL1: + case TWL6040_DAI_DL2: + twl6040_mute_path(dai->codec, dai->id, mute); + break; + default: + break; + } + + return 0; +} + static const struct snd_soc_dai_ops twl6040_dai_ops = { .startup = twl6040_startup, .hw_params = twl6040_hw_params, .prepare = twl6040_prepare, .set_sysclk = twl6040_set_dai_sysclk, + .digital_mute = twl6040_digital_mute, }; static struct snd_soc_dai_driver twl6040_dai[] = { -- cgit v1.2.1 From 7eebffd3f4328c6dc220521f14b384affdaf9427 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 24 Jun 2013 16:00:21 +0200 Subject: ALSA: hda - Add auto_mute_via_amp flag to generic parser Add a new flag, auto_mute_via_amp, to determine the behavior of the headphone / line-out auto-mute. When this flag is set, the generic driver mutes the speaker and line outputs via the amp mute of each pin, instead of changing the pin control values. This is introduced for devices that don't work expectedly with the pin control values; for example, some devices are known to keep enabling the speaker outputs no matter which pin control values are set on the speaker pins. The driver doesn't check actually whether the pins have the output amp caps, but assumes that the proper mixer (mute) controls are created on all these pins. If not the case, you can't use this flag for your device. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 47 ++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_generic.h | 4 ++++ 2 files changed, 50 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 4b1524a861f3..1485d871d628 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -133,6 +133,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "line_in_auto_switch"); if (val >= 0) spec->line_in_auto_switch = !!val; + val = snd_hda_get_bool_hint(codec, "auto_mute_via_amp"); + if (val >= 0) + spec->auto_mute_via_amp = !!val; val = snd_hda_get_bool_hint(codec, "need_dac_fix"); if (val >= 0) spec->need_dac_fix = !!val; @@ -808,6 +811,9 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx) * Helper functions for creating mixer ctl elements */ +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + enum { HDA_CTL_WIDGET_VOL, HDA_CTL_WIDGET_MUTE, @@ -815,7 +821,15 @@ enum { }; static const struct snd_kcontrol_new control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), + /* only the put callback is replaced for handling the special mute */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_AMP_FLAG, + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = hda_gen_mixer_mute_put, /* replaced */ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), + }, HDA_BIND_MUTE(NULL, 0, 0, 0), }; @@ -922,6 +936,23 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx, return add_sw_ctl(codec, pfx, cidx, chs, path); } +/* playback mute control with the software mute bit check */ +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gen_spec *spec = codec->spec; + + if (spec->auto_mute_via_amp) { + hda_nid_t nid = get_amp_nid(kcontrol); + bool enabled = !((spec->mute_bits >> nid) & 1); + ucontrol->value.integer.value[0] &= enabled; + ucontrol->value.integer.value[1] &= enabled; + } + + return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +} + /* any ctl assigned to the path with the given index? */ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) { @@ -3719,6 +3750,16 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, unsigned int val, oldval; if (!nid) break; + + if (spec->auto_mute_via_amp) { + if (mute) + spec->mute_bits |= (1ULL << nid); + else + spec->mute_bits &= ~(1ULL << nid); + set_pin_eapd(codec, nid, !mute); + continue; + } + oldval = snd_hda_codec_get_pin_target(codec, nid); if (oldval & PIN_IN) continue; /* no mute for inputs */ @@ -3786,6 +3827,10 @@ static void call_update_outputs(struct hda_codec *codec) spec->automute_hook(codec); else snd_hda_gen_update_outputs(codec); + + /* sync the whole vmaster slaves to reflect the new auto-mute status */ + if (spec->auto_mute_via_amp && !codec->bus->shutdown) + snd_ctl_sync_vmaster(spec->vmaster_mute.sw_kctl, false); } /* standard HP-automute helper */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 76200314ee95..e199a852388b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -209,6 +209,7 @@ struct hda_gen_spec { unsigned int master_mute:1; /* master mute over all */ unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */ unsigned int line_in_auto_switch:1; /* allow line-in auto switch */ + unsigned int auto_mute_via_amp:1; /* auto-mute via amp instead of pinctl */ /* parser behavior flags; set before snd_hda_gen_parse_auto_config() */ unsigned int suppress_auto_mute:1; /* suppress input jack auto mute */ @@ -237,6 +238,9 @@ struct hda_gen_spec { unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + /* additional mute flags (only effective with auto_mute_via_amp=1) */ + u64 mute_bits; + /* badness tables for output path evaluations */ const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; -- cgit v1.2.1 From eb33ccf7637c34b2c95dbcca8b2e4cab76a52949 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 24 Jun 2013 16:18:10 +0200 Subject: ALSA: hda - Use auto_mute_via_amp=1 for VT1708 We've got bug report wrt many machines with VT1708 (e.g. IBM POS machines) showing the broken auto-mute behavior. It turned out that the problem is that the pin control values of the speaker and line-out pins are completely ignored. As a workaround, let's use the newly introduced feature of the generic parser, to control the mute via amp on pins. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index dcebf3cb18de..e2481baddc70 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -741,6 +741,8 @@ static int patch_vt1708(struct hda_codec *codec) /* don't support the input jack switching due to lack of unsol event */ /* (it may work with polling, though, but it needs testing) */ spec->gen.suppress_auto_mic = 1; + /* Some machines show the broken speaker mute */ + spec->gen.auto_mute_via_amp = 1; /* Add HP and CD pin config connect bit re-config action */ vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID); -- cgit v1.2.1 From 8b2c7a5c404d7accb9790e1d5a1a518dd0a77a5e Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 24 Jun 2013 23:41:23 -0400 Subject: ALSA: hda - Add In-driver connection info Pin's connection list may change dynamically with hot-plug event on Intel Haswell chip. Users would see connections be "0" in codec#. when play audio on this pin, software driver choose converter from cache connections. So add "In-driver connection" info to avoid confuse when raw connections are different with cache connection. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0fee8fae590a..9760f001916d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -504,6 +504,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, int conn_len) { int c, curr = -1; + const hda_nid_t *list; + int cache_len; if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && @@ -521,6 +523,19 @@ static void print_conn_list(struct snd_info_buffer *buffer, } snd_iprintf(buffer, "\n"); } + + /* Get Cache connections info */ + cache_len = snd_hda_get_conn_list(codec, nid, &list); + if (cache_len != conn_len + || memcmp(list, conn, conn_len)) { + snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); + if (cache_len > 0) { + snd_iprintf(buffer, " "); + for (c = 0; c < cache_len; c++) + snd_iprintf(buffer, " 0x%02x", list[c]); + snd_iprintf(buffer, "\n"); + } + } } static void print_gpio(struct snd_info_buffer *buffer, -- cgit v1.2.1 From 7295b26438ec018a16159e45d514e1c94c554c5b Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 25 Jun 2013 05:58:49 -0400 Subject: ALSA: hda - clean up code to reset hda link This patch is a cleanup to the previous patch "reset hda link during system/ runtime suspend". In this patch - azx_enter_link_reset() and azx_exit_link_reset() are defined for entering and exiting the link reset respectively. azx_link_reset() is no longer used and replaced by azx_enter_link_reset(). - azx_reset() reuses the above two new functions for a link reset cycle Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 35 ++++++++++++++++++----------------- 1 file changed, 18 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f110c7ba092..f39de9055097 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1121,7 +1121,7 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, #endif /* enter link reset */ -static void azx_reset_link(struct azx *chip) +static void azx_enter_link_reset(struct azx *chip) { unsigned long timeout; @@ -1134,11 +1134,22 @@ static void azx_reset_link(struct azx *chip) usleep_range(500, 1000); } -/* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +/* exit link reset */ +static void azx_exit_link_reset(struct azx *chip) { unsigned long timeout; + azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + + timeout = jiffies + msecs_to_jiffies(100); + while (!azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); +} + +/* reset codec link */ +static int azx_reset(struct azx *chip, int full_reset) +{ if (!full_reset) goto __skip; @@ -1146,12 +1157,7 @@ static int azx_reset(struct azx *chip, int full_reset) azx_writeb(chip, STATESTS, STATESTS_INT_MASK); /* reset controller */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_enter_link_reset(chip); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 @@ -1159,12 +1165,7 @@ static int azx_reset(struct azx *chip, int full_reset) usleep_range(500, 1000); /* Bring controller out of reset */ - azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (!azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_exit_link_reset(chip); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ usleep_range(1000, 1200); @@ -2908,7 +2909,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2961,7 +2962,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); return 0; } -- cgit v1.2.1 From 28cb72e5b86bb8340568c2ceb940eb165a9791b3 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 24 Jun 2013 07:45:23 -0400 Subject: ALSA: hda/hdmi - poll eld at resume time Hdmi driver may not receive intrinsic event from gfx side when it's in runtime suspend mode. There's no ELD info when exit from runtime suspend. This patch avoid missing ELD info. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8428763de153..540bdef2f904 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1864,12 +1864,33 @@ static void generic_hdmi_free(struct hda_codec *codec) kfree(spec); } +#ifdef CONFIG_PM +static int generic_hdmi_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + generic_hdmi_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hdmi_present_sense(per_pin, 1); + } + return 0; +} +#endif + static const struct hda_codec_ops generic_hdmi_patch_ops = { .init = generic_hdmi_init, .free = generic_hdmi_free, .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, +#ifdef CONFIG_PM + .resume = generic_hdmi_resume, +#endif }; -- cgit v1.2.1 From 58e22201f8a4c270300c589083ff9117b3b8274c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Jun 2013 09:27:19 +0200 Subject: ALSA: hda - Remove superfluous stac_resume() The stac_resume() is exactly what the default resume code does, so we don't have to define and use it doubly. Let's cut it off. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9b6cb270dbe5..e2f83591161b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3711,14 +3711,6 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #endif #ifdef CONFIG_PM -static int stac_resume(struct hda_codec *codec) -{ - codec->patch_ops.init(codec); - snd_hda_codec_resume_amp(codec); - snd_hda_codec_resume_cache(codec); - return 0; -} - static int stac_suspend(struct hda_codec *codec) { stac_shutup(codec); @@ -3747,7 +3739,6 @@ static void stac_set_power_state(struct hda_codec *codec, hda_nid_t fg, } #else #define stac_suspend NULL -#define stac_resume NULL #define stac_set_power_state NULL #endif /* CONFIG_PM */ @@ -3759,7 +3750,6 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, - .resume = stac_resume, #endif .reboot_notify = stac_shutup, }; -- cgit v1.2.1 From 0623a889d1c2eb6219576b13263bcd24133c971a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Jun 2013 09:28:40 +0200 Subject: ALSA: hda - Add missing alc_inv_dmic_sync() call in alc269_resume() As some of ALC269 quirks use the inverted dmic feature, we need to call alc_inv_dmic_sync() in the resume callback like in alc_resume(), too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ad087ea32f3a..ae121113f223 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2603,6 +2603,7 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); + alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); return 0; } -- cgit v1.2.1 From f02fe86199d90cd3c7c0c51e223840c3891e5d18 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 23 Jun 2013 11:50:24 +0900 Subject: ALSA: snd-firewire-lib: remove unused header inclusion spinlock is not used in amdtp.h. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.h | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index b680c5ef01d6..f6103d68c4b1 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -3,7 +3,6 @@ #include #include -#include #include "packets-buffer.h" /** -- cgit v1.2.1 From 6b36d370ad66aa73328a0cd8763f6028e7b28f6c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:25:29 +0200 Subject: ASoC: tas5086: open-code I2C transfer routines In order to support registers of unequal sizes, the I2C I/O has to be open-coded. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 85 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 84 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index d447c4aa1d5e..57c9de02b14f 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -119,6 +119,17 @@ static const struct reg_default tas5086_reg_defaults[] = { { 0x1c, 0x05 }, }; +static int tas5086_register_size(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5086_DEV_ID ... TAS5086_BKNDERR: + return 1; + } + + dev_err(dev, "Unsupported register address: %d\n", reg); + return 0; +} + static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) { return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); @@ -140,6 +151,76 @@ static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); } +static int tas5086_reg_write(void *context, unsigned int reg, + unsigned int value) +{ + struct i2c_client *client = context; + unsigned int i, size; + uint8_t buf[5]; + int ret; + + size = tas5086_register_size(&client->dev, reg); + if (size == 0) + return -EINVAL; + + buf[0] = reg; + + for (i = size; i >= 1; --i) { + buf[i] = value; + value >>= 8; + } + + ret = i2c_master_send(client, buf, size + 1); + if (ret == size + 1) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static int tas5086_reg_read(void *context, unsigned int reg, + unsigned int *value) +{ + struct i2c_client *client = context; + uint8_t send_buf, recv_buf[4]; + struct i2c_msg msgs[2]; + unsigned int size; + unsigned int i; + int ret; + + size = tas5086_register_size(&client->dev, reg); + if (size == 0) + return -EINVAL; + + send_buf = reg; + + msgs[0].addr = client->addr; + msgs[0].len = sizeof(send_buf); + msgs[0].buf = &send_buf; + msgs[0].flags = 0; + + msgs[1].addr = client->addr; + msgs[1].len = size; + msgs[1].buf = recv_buf; + msgs[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs)); + if (ret < 0) + return ret; + else if (ret != ARRAY_SIZE(msgs)) + return -EIO; + + *value = 0; + + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } + + return 0; +} + struct tas5086_private { struct regmap *regmap; unsigned int mclk, sclk; @@ -508,6 +589,8 @@ static const struct regmap_config tas5086_regmap = { .volatile_reg = tas5086_volatile_reg, .writeable_reg = tas5086_writeable_reg, .readable_reg = tas5086_accessible_reg, + .reg_read = tas5086_reg_read, + .reg_write = tas5086_reg_write, }; static int tas5086_i2c_probe(struct i2c_client *i2c, @@ -522,7 +605,7 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; - priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); -- cgit v1.2.1 From 8892d479f1ba505c835e31bb66fcf3994f5127aa Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:25:30 +0200 Subject: ASoC: tas5086: add more register defines Add register definitions for input and output mux registers, and rewrite the tas5086_accessible_reg() function. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 57c9de02b14f..8130ab5d9848 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -83,6 +83,10 @@ #define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ #define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ #define TAS5086_BKNDERR 0x1c +#define TAS5086_INPUT_MUX 0x20 +#define TAS5086_PWM_OUTPUT_MUX 0x25 + +#define TAS5086_MAX_REGISTER TAS5086_PWM_OUTPUT_MUX /* * Default TAS5086 power-up configuration @@ -124,6 +128,9 @@ static int tas5086_register_size(struct device *dev, unsigned int reg) switch (reg) { case TAS5086_DEV_ID ... TAS5086_BKNDERR: return 1; + case TAS5086_INPUT_MUX: + case TAS5086_PWM_OUTPUT_MUX: + return 4; } dev_err(dev, "Unsupported register address: %d\n", reg); @@ -132,7 +139,14 @@ static int tas5086_register_size(struct device *dev, unsigned int reg) static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) { - return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); + switch (reg) { + case 0x0f: + case 0x11 ... 0x17: + case 0x1d ... 0x1f: + return false; + default: + return true; + } } static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) @@ -581,8 +595,8 @@ MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); static const struct regmap_config tas5086_regmap = { .reg_bits = 8, - .val_bits = 8, - .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .val_bits = 32, + .max_register = TAS5086_MAX_REGISTER, .reg_defaults = tas5086_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), .cache_type = REGCACHE_RBTREE, -- cgit v1.2.1 From 18710acdeea02777ef70013465f6f7fced411096 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:25:31 +0200 Subject: ASoC: tas5086: add DAPM mux controls The TAS5086 has two muxes, one for connecting I2S inputs to internal channels, and another one for selecting which internal channel should be routed to which PWM output pin. This patch adds DAPM widgets and routes for this driver. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 200 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 200 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 8130ab5d9848..bcbbec1399b8 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -471,6 +471,202 @@ static const struct snd_kcontrol_new tas5086_controls[] = { tas5086_get_deemph, tas5086_put_deemph), }; +/* Input mux controls */ +static const char *tas5086_dapm_sdin_texts[] = +{ + "SDIN1-L", "SDIN1-R", "SDIN2-L", "SDIN2-R", + "SDIN3-L", "SDIN3-R", "Ground (0)", "nc" +}; + +static const struct soc_enum tas5086_dapm_input_mux_enum[] = { + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 20, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 16, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 12, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 8, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 4, 8, tas5086_dapm_sdin_texts), + SOC_ENUM_SINGLE(TAS5086_INPUT_MUX, 0, 8, tas5086_dapm_sdin_texts), +}; + +static const struct snd_kcontrol_new tas5086_dapm_input_mux_controls[] = { + SOC_DAPM_ENUM("Channel 1 input", tas5086_dapm_input_mux_enum[0]), + SOC_DAPM_ENUM("Channel 2 input", tas5086_dapm_input_mux_enum[1]), + SOC_DAPM_ENUM("Channel 3 input", tas5086_dapm_input_mux_enum[2]), + SOC_DAPM_ENUM("Channel 4 input", tas5086_dapm_input_mux_enum[3]), + SOC_DAPM_ENUM("Channel 5 input", tas5086_dapm_input_mux_enum[4]), + SOC_DAPM_ENUM("Channel 6 input", tas5086_dapm_input_mux_enum[5]), +}; + +/* Output mux controls */ +static const char *tas5086_dapm_channel_texts[] = + { "Channel 1 Mux", "Channel 2 Mux", "Channel 3 Mux", + "Channel 4 Mux", "Channel 5 Mux", "Channel 6 Mux" }; + +static const struct soc_enum tas5086_dapm_output_mux_enum[] = { + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 20, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 16, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 12, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 8, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 4, 6, tas5086_dapm_channel_texts), + SOC_ENUM_SINGLE(TAS5086_PWM_OUTPUT_MUX, 0, 6, tas5086_dapm_channel_texts), +}; + +static const struct snd_kcontrol_new tas5086_dapm_output_mux_controls[] = { + SOC_DAPM_ENUM("PWM1 Output", tas5086_dapm_output_mux_enum[0]), + SOC_DAPM_ENUM("PWM2 Output", tas5086_dapm_output_mux_enum[1]), + SOC_DAPM_ENUM("PWM3 Output", tas5086_dapm_output_mux_enum[2]), + SOC_DAPM_ENUM("PWM4 Output", tas5086_dapm_output_mux_enum[3]), + SOC_DAPM_ENUM("PWM5 Output", tas5086_dapm_output_mux_enum[4]), + SOC_DAPM_ENUM("PWM6 Output", tas5086_dapm_output_mux_enum[5]), +}; + +static const struct snd_soc_dapm_widget tas5086_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("SDIN1-L"), + SND_SOC_DAPM_INPUT("SDIN1-R"), + SND_SOC_DAPM_INPUT("SDIN2-L"), + SND_SOC_DAPM_INPUT("SDIN2-R"), + SND_SOC_DAPM_INPUT("SDIN3-L"), + SND_SOC_DAPM_INPUT("SDIN3-R"), + SND_SOC_DAPM_INPUT("SDIN4-L"), + SND_SOC_DAPM_INPUT("SDIN4-R"), + + SND_SOC_DAPM_OUTPUT("PWM1"), + SND_SOC_DAPM_OUTPUT("PWM2"), + SND_SOC_DAPM_OUTPUT("PWM3"), + SND_SOC_DAPM_OUTPUT("PWM4"), + SND_SOC_DAPM_OUTPUT("PWM5"), + SND_SOC_DAPM_OUTPUT("PWM6"), + + SND_SOC_DAPM_MUX("Channel 1 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[0]), + SND_SOC_DAPM_MUX("Channel 2 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[1]), + SND_SOC_DAPM_MUX("Channel 3 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[2]), + SND_SOC_DAPM_MUX("Channel 4 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[3]), + SND_SOC_DAPM_MUX("Channel 5 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[4]), + SND_SOC_DAPM_MUX("Channel 6 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_input_mux_controls[5]), + + SND_SOC_DAPM_MUX("PWM1 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[0]), + SND_SOC_DAPM_MUX("PWM2 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[1]), + SND_SOC_DAPM_MUX("PWM3 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[2]), + SND_SOC_DAPM_MUX("PWM4 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[3]), + SND_SOC_DAPM_MUX("PWM5 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[4]), + SND_SOC_DAPM_MUX("PWM6 Mux", SND_SOC_NOPM, 0, 0, + &tas5086_dapm_output_mux_controls[5]), +}; + +static const struct snd_soc_dapm_route tas5086_dapm_routes[] = { + /* SDIN inputs -> channel muxes */ + { "Channel 1 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 1 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 1 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 1 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 1 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 1 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 2 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 2 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 2 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 2 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 2 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 2 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 2 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 2 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 2 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 2 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 2 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 2 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 3 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 3 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 3 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 3 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 3 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 3 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 4 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 4 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 4 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 4 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 4 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 4 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 5 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 5 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 5 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 5 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 5 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 5 Mux", "SDIN3-R", "SDIN3-R" }, + + { "Channel 6 Mux", "SDIN1-L", "SDIN1-L" }, + { "Channel 6 Mux", "SDIN1-R", "SDIN1-R" }, + { "Channel 6 Mux", "SDIN2-L", "SDIN2-L" }, + { "Channel 6 Mux", "SDIN2-R", "SDIN2-R" }, + { "Channel 6 Mux", "SDIN3-L", "SDIN3-L" }, + { "Channel 6 Mux", "SDIN3-R", "SDIN3-R" }, + + /* Channel muxes -> PWM muxes */ + { "PWM1 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM2 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM3 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM4 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM5 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + { "PWM6 Mux", "Channel 1 Mux", "Channel 1 Mux" }, + + { "PWM1 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM2 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM3 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM4 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM5 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + { "PWM6 Mux", "Channel 2 Mux", "Channel 2 Mux" }, + + { "PWM1 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM2 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM3 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM4 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM5 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + { "PWM6 Mux", "Channel 3 Mux", "Channel 3 Mux" }, + + { "PWM1 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM2 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM3 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM4 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM5 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + { "PWM6 Mux", "Channel 4 Mux", "Channel 4 Mux" }, + + { "PWM1 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM2 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM3 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM4 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM5 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + { "PWM6 Mux", "Channel 5 Mux", "Channel 5 Mux" }, + + { "PWM1 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM2 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM3 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM4 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM5 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + { "PWM6 Mux", "Channel 6 Mux", "Channel 6 Mux" }, + + /* The PWM muxes are directly connected to the PWM outputs */ + { "PWM1", NULL, "PWM1 Mux" }, + { "PWM2", NULL, "PWM2 Mux" }, + { "PWM3", NULL, "PWM3 Mux" }, + { "PWM4", NULL, "PWM4 Mux" }, + { "PWM5", NULL, "PWM5 Mux" }, + { "PWM6", NULL, "PWM6 Mux" }, + +}; + static const struct snd_soc_dai_ops tas5086_dai_ops = { .hw_params = tas5086_hw_params, .set_sysclk = tas5086_set_dai_sysclk, @@ -585,6 +781,10 @@ static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { .resume = tas5086_soc_resume, .controls = tas5086_controls, .num_controls = ARRAY_SIZE(tas5086_controls), + .dapm_widgets = tas5086_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5086_dapm_widgets), + .dapm_routes = tas5086_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tas5086_dapm_routes), }; static const struct i2c_device_id tas5086_i2c_id[] = { -- cgit v1.2.1 From 79b23b564060c5483a489562b01a6eb778a312f7 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:25:32 +0200 Subject: ASoC: tas5086: add support for pwm start mode config The TAS5086 has two alternative modes to start its PWM channels, Mid-Z and Low-Z. Which one to use depends on how the PWM power stages are connected to the TAS5086. This patch adds 6 optional boolean properties to the DT bindings of the driver which allow the user to configure each individual channel to the Mid-Z scheme, and leaves all the others to the default (Low-Z). Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index bcbbec1399b8..72067f79633e 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -88,6 +88,10 @@ #define TAS5086_MAX_REGISTER TAS5086_PWM_OUTPUT_MUX +#define TAS5086_PWM_START_MIDZ_FOR_START_1 (1 << 7) +#define TAS5086_PWM_START_MIDZ_FOR_START_2 (1 << 6) +#define TAS5086_PWM_START_CHANNEL_MASK (0x3f) + /* * Default TAS5086 power-up configuration */ @@ -717,13 +721,31 @@ static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int charge_period = 1300000; /* hardware default is 1300 ms */ + u8 pwm_start = TAS5086_PWM_START_CHANNEL_MASK; int i, ret; if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; of_property_read_u32(of_node, "ti,charge-period", &charge_period); + + for (i = 0; i < 6; i++) { + char name[25]; + + snprintf(name, sizeof(name), + "ti,mid-z-channel-%d", i + 1); + + if (of_get_property(of_node, name, NULL) != NULL) + pwm_start &= ~(1 << i); + } } + /* + * Configure 'part 2' of the PWM starts to always use MID-Z, and tell + * all configured mid-z channels to start start under 'part 2'. + */ + regmap_write(priv->regmap, TAS5086_PWM_START, + TAS5086_PWM_START_MIDZ_FOR_START_2 | pwm_start); + /* lookup and set split-capacitor charge period */ if (charge_period == 0) { regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); -- cgit v1.2.1 From de9fc724daaf5ceaf0af6ef23b2b3b1d933273e3 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:31:29 +0200 Subject: ASoC: adau1701: move firmware download to adau1701_reset() The chip needs a new download after each reset, so the code to do that needs to live in adau1701_reset(). Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 32 ++++++++++++-------------------- 1 file changed, 12 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index b6b1a773bd37..997fc3b881fe 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -184,27 +184,20 @@ static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) return value; } -static void adau1701_reset(struct snd_soc_codec *codec) +static int adau1701_reset(struct snd_soc_codec *codec) { struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); - - if (!gpio_is_valid(adau1701->gpio_nreset)) - return; - - gpio_set_value(adau1701->gpio_nreset, 0); - /* minimum reset time is 20ns */ - udelay(1); - gpio_set_value(adau1701->gpio_nreset, 1); - /* power-up time may be as long as 85ms */ - mdelay(85); -} - -static int adau1701_init(struct snd_soc_codec *codec) -{ - int ret; struct i2c_client *client = to_i2c_client(codec->dev); + int ret; - adau1701_reset(codec); + if (gpio_is_valid(adau1701->gpio_nreset)) { + gpio_set_value(adau1701->gpio_nreset, 0); + /* minimum reset time is 20ns */ + udelay(1); + gpio_set_value(adau1701->gpio_nreset, 1); + /* power-up time may be as long as 85ms */ + mdelay(85); + } ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); if (ret) { @@ -213,6 +206,7 @@ static int adau1701_init(struct snd_soc_codec *codec) } snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); return 0; } @@ -498,12 +492,10 @@ static int adau1701_probe(struct snd_soc_codec *codec) codec->control_data = to_i2c_client(codec->dev); - ret = adau1701_init(codec); + ret = adau1701_reset(codec); if (ret) return ret; - snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); - return 0; } -- cgit v1.2.1 From 2352d4bf43b105ec2da5f43211db4a4c9bf34d4e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:31:30 +0200 Subject: ASoC: adau1701: allow configuration of PLL mode pins The ADAU1701 has 2 hardware pins to configure the PLL mode in accordance to the MCLK-to-LRCLK ratio. These pins have to be stable before the chip is released from reset, and a full reset cycle, including a new firmware download is needed whenever they change. This patch adds GPIO properties to the DT bindings of the Codec, and implements makes the set_sysclk memorize the configured sysclk. Because the run-time parameters are unknown at probe time, the first firmware download is postponed to the first hw_params call, when the driver can determine the mclk/lrclk divider. Subsequent downloads are only issued when the divider configuration changes. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 105 +++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 98 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 997fc3b881fe..770d90ee5f92 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -87,11 +87,16 @@ #define ADAU1701_OSCIPOW_OPD 0x04 #define ADAU1701_DACSET_DACINIT 1 +#define ADAU1707_CLKDIV_UNSET (-1UL) + #define ADAU1701_FIRMWARE "adau1701.bin" struct adau1701 { int gpio_nreset; + int gpio_pll_mode[2]; unsigned int dai_fmt; + unsigned int pll_clkdiv; + unsigned int sysclk; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -184,12 +189,38 @@ static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) return value; } -static int adau1701_reset(struct snd_soc_codec *codec) +static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) { struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); struct i2c_client *client = to_i2c_client(codec->dev); int ret; + if (clkdiv != ADAU1707_CLKDIV_UNSET && + gpio_is_valid(adau1701->gpio_pll_mode[0]) && + gpio_is_valid(adau1701->gpio_pll_mode[1])) { + switch (clkdiv) { + case 64: + gpio_set_value(adau1701->gpio_pll_mode[0], 0); + gpio_set_value(adau1701->gpio_pll_mode[1], 0); + break; + case 256: + gpio_set_value(adau1701->gpio_pll_mode[0], 0); + gpio_set_value(adau1701->gpio_pll_mode[1], 1); + break; + case 384: + gpio_set_value(adau1701->gpio_pll_mode[0], 1); + gpio_set_value(adau1701->gpio_pll_mode[1], 0); + break; + case 0: /* fallback */ + case 512: + gpio_set_value(adau1701->gpio_pll_mode[0], 1); + gpio_set_value(adau1701->gpio_pll_mode[1], 1); + break; + } + } + + adau1701->pll_clkdiv = clkdiv; + if (gpio_is_valid(adau1701->gpio_nreset)) { gpio_set_value(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ @@ -199,10 +230,16 @@ static int adau1701_reset(struct snd_soc_codec *codec) mdelay(85); } - ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); - if (ret) { - dev_warn(codec->dev, "Failed to load firmware\n"); - return ret; + /* + * Postpone the firmware download to a point in time when we + * know the correct PLL setup + */ + if (clkdiv != ADAU1707_CLKDIV_UNSET) { + ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); + if (ret) { + dev_warn(codec->dev, "Failed to load firmware\n"); + return ret; + } } snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); @@ -285,8 +322,22 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int clkdiv = adau1701->sysclk / params_rate(params); snd_pcm_format_t format; unsigned int val; + int ret; + + /* + * If the mclk/lrclk ratio changes, the chip needs updated PLL + * mode GPIO settings, and a full reset cycle, including a new + * firmware upload. + */ + if (clkdiv != adau1701->pll_clkdiv) { + ret = adau1701_reset(codec, clkdiv); + if (ret < 0) + return ret; + } switch (params_rate(params)) { case 192000: @@ -429,6 +480,7 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { unsigned int val; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (clk_id) { case ADAU1701_CLK_SRC_OSC: @@ -442,6 +494,7 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, } snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + adau1701->sysclk = freq; return 0; } @@ -489,11 +542,21 @@ MODULE_DEVICE_TABLE(of, adau1701_dt_ids); static int adau1701_probe(struct snd_soc_codec *codec) { int ret; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); codec->control_data = to_i2c_client(codec->dev); - ret = adau1701_reset(codec); - if (ret) + /* + * Let the pll_clkdiv variable default to something that won't happen + * at runtime. That way, we can postpone the firmware download from + * adau1701_reset() to a point in time when we know the correct PLL + * mode parameters. + */ + adau1701->pll_clkdiv = ADAU1707_CLKDIV_UNSET; + + /* initalize with pre-configured pll mode settings */ + ret = adau1701_reset(codec, adau1701->pll_clkdiv); + if (ret < 0) return ret; return 0; @@ -526,6 +589,7 @@ static int adau1701_i2c_probe(struct i2c_client *client, struct adau1701 *adau1701; struct device *dev = &client->dev; int gpio_nreset = -EINVAL; + int gpio_pll_mode[2] = { -EINVAL, -EINVAL }; int ret; adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); @@ -536,6 +600,16 @@ static int adau1701_i2c_probe(struct i2c_client *client, gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); if (gpio_nreset < 0 && gpio_nreset != -ENOENT) return gpio_nreset; + + gpio_pll_mode[0] = of_get_named_gpio(dev->of_node, + "adi,pll-mode-gpios", 0); + if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) + return gpio_pll_mode[0]; + + gpio_pll_mode[1] = of_get_named_gpio(dev->of_node, + "adi,pll-mode-gpios", 1); + if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) + return gpio_pll_mode[1]; } if (gpio_is_valid(gpio_nreset)) { @@ -545,7 +619,24 @@ static int adau1701_i2c_probe(struct i2c_client *client, return ret; } + if (gpio_is_valid(gpio_pll_mode[0]) && + gpio_is_valid(gpio_pll_mode[1])) { + ret = devm_gpio_request_one(dev, gpio_pll_mode[0], + GPIOF_OUT_INIT_LOW, + "ADAU1701 PLL mode 0"); + if (ret < 0) + return ret; + + ret = devm_gpio_request_one(dev, gpio_pll_mode[1], + GPIOF_OUT_INIT_LOW, + "ADAU1701 PLL mode 1"); + if (ret < 0) + return ret; + } + adau1701->gpio_nreset = gpio_nreset; + adau1701->gpio_pll_mode[0] = gpio_pll_mode[0]; + adau1701->gpio_pll_mode[1] = gpio_pll_mode[1]; i2c_set_clientdata(client, adau1701); ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, -- cgit v1.2.1 From 45405d58924f49a13b924ed6db6fe47981487b4a Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:31:31 +0200 Subject: ASoC: adau1701: switch to direct regmap API usage The hardware I/O has to be open-coded due to registers of unequal sizes. Other than that, the transition is straight forward. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 118 +++++++++++++++++++++++++++++++------------- 1 file changed, 85 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 770d90ee5f92..881bab4a65aa 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -24,16 +25,16 @@ #include "sigmadsp.h" #include "adau1701.h" -#define ADAU1701_DSPCTRL 0x1c -#define ADAU1701_SEROCTL 0x1e -#define ADAU1701_SERICTL 0x1f +#define ADAU1701_DSPCTRL 0x081c +#define ADAU1701_SEROCTL 0x081e +#define ADAU1701_SERICTL 0x081f -#define ADAU1701_AUXNPOW 0x22 +#define ADAU1701_AUXNPOW 0x0822 -#define ADAU1701_OSCIPOW 0x26 -#define ADAU1701_DACSET 0x27 +#define ADAU1701_OSCIPOW 0x0826 +#define ADAU1701_DACSET 0x0827 -#define ADAU1701_NUM_REGS 0x28 +#define ADAU1701_MAX_REGISTER 0x0828 #define ADAU1701_DSPCTRL_CR (1 << 2) #define ADAU1701_DSPCTRL_DAM (1 << 3) @@ -97,6 +98,7 @@ struct adau1701 { unsigned int dai_fmt; unsigned int pll_clkdiv; unsigned int sysclk; + struct regmap *regmap; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -128,7 +130,7 @@ static const struct snd_soc_dapm_route adau1701_dapm_routes[] = { { "ADC", NULL, "IN1" }, }; -static unsigned int adau1701_register_size(struct snd_soc_codec *codec, +static unsigned int adau1701_register_size(struct device *dev, unsigned int reg) { switch (reg) { @@ -142,33 +144,42 @@ static unsigned int adau1701_register_size(struct snd_soc_codec *codec, return 1; } - dev_err(codec->dev, "Unsupported register address: %d\n", reg); + dev_err(dev, "Unsupported register address: %d\n", reg); return 0; } -static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) +static bool adau1701_volatile_reg(struct device *dev, unsigned int reg) { + switch (reg) { + case ADAU1701_DACSET: + return true; + default: + return false; + } +} + +static int adau1701_reg_write(void *context, unsigned int reg, + unsigned int value) +{ + struct i2c_client *client = context; unsigned int i; unsigned int size; uint8_t buf[4]; int ret; - size = adau1701_register_size(codec, reg); + size = adau1701_register_size(&client->dev, reg); if (size == 0) return -EINVAL; - snd_soc_cache_write(codec, reg, value); - - buf[0] = 0x08; - buf[1] = reg; + buf[0] = reg >> 8; + buf[1] = reg & 0xff; for (i = size + 1; i >= 2; --i) { buf[i] = value; value >>= 8; } - ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2); + ret = i2c_master_send(client, buf, size + 2); if (ret == size + 2) return 0; else if (ret < 0) @@ -177,16 +188,45 @@ static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; } -static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) +static int adau1701_reg_read(void *context, unsigned int reg, + unsigned int *value) { - unsigned int value; - unsigned int ret; + int ret; + unsigned int i; + unsigned int size; + uint8_t send_buf[2], recv_buf[3]; + struct i2c_client *client = context; + struct i2c_msg msgs[2]; + + size = adau1701_register_size(&client->dev, reg); + if (size == 0) + return -EINVAL; - ret = snd_soc_cache_read(codec, reg, &value); - if (ret) + send_buf[0] = reg >> 8; + send_buf[1] = reg & 0xff; + + msgs[0].addr = client->addr; + msgs[0].len = sizeof(send_buf); + msgs[0].buf = send_buf; + msgs[0].flags = 0; + + msgs[1].addr = client->addr; + msgs[1].len = size; + msgs[1].buf = recv_buf; + msgs[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs)); + if (ret < 0) return ret; + else if (ret != ARRAY_SIZE(msgs)) + return -EIO; + + *value = 0; + + for (i = 0; i < size; i++) + *value |= recv_buf[i] << (i * 8); - return value; + return 0; } static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) @@ -242,8 +282,11 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) } } - snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); - snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + regmap_write(adau1701->regmap, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + regmap_write(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + + regcache_mark_dirty(adau1701->regmap); + regcache_sync(adau1701->regmap); return 0; } @@ -429,8 +472,8 @@ static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai, adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - snd_soc_write(codec, ADAU1701_SERICTL, serictl); - snd_soc_update_bits(codec, ADAU1701_SEROCTL, + regmap_write(adau1701->regmap, ADAU1701_SERICTL, serictl); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl); return 0; @@ -567,9 +610,6 @@ static struct snd_soc_codec_driver adau1701_codec_drv = { .set_bias_level = adau1701_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ADAU1701_NUM_REGS, - .reg_word_size = sizeof(u16), - .controls = adau1701_controls, .num_controls = ARRAY_SIZE(adau1701_controls), .dapm_widgets = adau1701_dapm_widgets, @@ -577,12 +617,19 @@ static struct snd_soc_codec_driver adau1701_codec_drv = { .dapm_routes = adau1701_dapm_routes, .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes), - .write = adau1701_write, - .read = adau1701_read, - .set_sysclk = adau1701_set_sysclk, }; +static const struct regmap_config adau1701_regmap = { + .reg_bits = 16, + .val_bits = 32, + .max_register = ADAU1701_MAX_REGISTER, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = adau1701_volatile_reg, + .reg_write = adau1701_reg_write, + .reg_read = adau1701_reg_read, +}; + static int adau1701_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -596,6 +643,11 @@ static int adau1701_i2c_probe(struct i2c_client *client, if (!adau1701) return -ENOMEM; + adau1701->regmap = devm_regmap_init(dev, NULL, client, + &adau1701_regmap); + if (IS_ERR(adau1701->regmap)) + return PTR_ERR(adau1701->regmap); + if (dev->of_node) { gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); if (gpio_nreset < 0 && gpio_nreset != -ENOENT) -- cgit v1.2.1 From 97d0a868450d08fae0db3f53459852901c6e2f4f Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 24 Jun 2013 16:31:32 +0200 Subject: ASoC: adau1701: add support for pin muxing The ADAU1701 has 12 pins that can be configured depending on the system configuration. Allow settting the corresponding registers from DT. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 881bab4a65aa..0e250f118c0e 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -29,6 +29,9 @@ #define ADAU1701_SEROCTL 0x081e #define ADAU1701_SERICTL 0x081f +#define ADAU1701_AUXNPOW 0x0822 +#define ADAU1701_PINCONF_0 0x0820 +#define ADAU1701_PINCONF_1 0x0821 #define ADAU1701_AUXNPOW 0x0822 #define ADAU1701_OSCIPOW 0x0826 @@ -99,6 +102,7 @@ struct adau1701 { unsigned int pll_clkdiv; unsigned int sysclk; struct regmap *regmap; + u8 pin_config[12]; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -134,6 +138,9 @@ static unsigned int adau1701_register_size(struct device *dev, unsigned int reg) { switch (reg) { + case ADAU1701_PINCONF_0: + case ADAU1701_PINCONF_1: + return 3; case ADAU1701_DSPCTRL: case ADAU1701_SEROCTL: case ADAU1701_AUXNPOW: @@ -164,7 +171,7 @@ static int adau1701_reg_write(void *context, unsigned int reg, struct i2c_client *client = context; unsigned int i; unsigned int size; - uint8_t buf[4]; + uint8_t buf[5]; int ret; size = adau1701_register_size(&client->dev, reg); @@ -584,7 +591,8 @@ MODULE_DEVICE_TABLE(of, adau1701_dt_ids); static int adau1701_probe(struct snd_soc_codec *codec) { - int ret; + int i, ret; + unsigned int val; struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); codec->control_data = to_i2c_client(codec->dev); @@ -602,6 +610,19 @@ static int adau1701_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; + /* set up pin config */ + val = 0; + for (i = 0; i < 6; i++) + val |= adau1701->pin_config[i] << (i * 4); + + regmap_write(adau1701->regmap, ADAU1701_PINCONF_0, val); + + val = 0; + for (i = 0; i < 6; i++) + val |= adau1701->pin_config[i + 6] << (i * 4); + + regmap_write(adau1701->regmap, ADAU1701_PINCONF_1, val); + return 0; } @@ -662,6 +683,13 @@ static int adau1701_i2c_probe(struct i2c_client *client, "adi,pll-mode-gpios", 1); if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) return gpio_pll_mode[1]; + + of_property_read_u32(dev->of_node, "adi,pll-clkdiv", + &adau1701->pll_clkdiv); + + of_property_read_u8_array(dev->of_node, "adi,pin-config", + adau1701->pin_config, + ARRAY_SIZE(adau1701->pin_config)); } if (gpio_is_valid(gpio_nreset)) { -- cgit v1.2.1 From 4bf07eef016e606a73ecae9762e155e51c5a38ed Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 25 Jun 2013 17:06:15 +0100 Subject: ASoC: stac9766: Remove version number There is no need to have versioning beyond that for the kernel, especially when the version number never gets updated. Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 2eda85ba79ac..cbc7ae322324 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -28,8 +28,6 @@ #include "stac9766.h" -#define STAC9766_VERSION "0.10" - /* * STAC9766 register cache */ @@ -338,8 +336,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; -- cgit v1.2.1 From bd5fe738e388ceaa32e5171481e0d3ec59f0ccfe Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 26 Jun 2013 10:52:20 +0300 Subject: ALSA: ak4xx-adda: info leak in ak4xxx_capture_source_info() "idx" is controled by the user and can be a negative offset into the input_names[] array. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4xxx-adda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index cef813d23641..ed726d1569e8 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -571,7 +571,7 @@ static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol); int mixer_ch = AK_GET_SHIFT(kcontrol->private_value); const char **input_names; - int num_names, idx; + unsigned int num_names, idx; num_names = ak4xxx_capture_num_inputs(ak, mixer_ch); if (!num_names) -- cgit v1.2.1 From de693006c9c813171e8d3124ee6afe4318cb9915 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 25 Jun 2013 10:16:56 +0800 Subject: ASoC: mid-x86: Convert to use devm_* APIs devm_* APIs are device managed and make code simpler. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 78d582519891..aec29a805354 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -371,7 +371,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) /* audio interrupt base of SRAM location where * interrupts are stored by System FW */ - mc_drv_ctx = kzalloc(sizeof(*mc_drv_ctx), GFP_ATOMIC); + mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); if (!mc_drv_ctx) { pr_err("allocation failed\n"); return -ENOMEM; @@ -381,40 +381,33 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) pdev, IORESOURCE_MEM, "IRQ_BASE"); if (!irq_mem) { pr_err("no mem resource given\n"); - ret_val = -ENODEV; - goto unalloc; + return -ENODEV; } - mc_drv_ctx->int_base = ioremap_nocache(irq_mem->start, - resource_size(irq_mem)); + mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, + resource_size(irq_mem)); if (!mc_drv_ctx->int_base) { pr_err("Mapping of cache failed\n"); - ret_val = -ENOMEM; - goto unalloc; + return -ENOMEM; } /* register for interrupt */ - ret_val = request_threaded_irq(irq, snd_mfld_jack_intr_handler, + ret_val = devm_request_threaded_irq(&pdev->dev, irq, + snd_mfld_jack_intr_handler, snd_mfld_jack_detection, IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); if (ret_val) { pr_err("cannot register IRQ\n"); - goto unalloc; + return ret_val; } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; ret_val = snd_soc_register_card(&snd_soc_card_mfld); if (ret_val) { pr_debug("snd_soc_register_card failed %d\n", ret_val); - goto freeirq; + return ret_val; } platform_set_drvdata(pdev, mc_drv_ctx); pr_debug("successfully exited probe\n"); - return ret_val; - -freeirq: - free_irq(irq, mc_drv_ctx); -unalloc: - kfree(mc_drv_ctx); - return ret_val; + return 0; } static int snd_mfld_mc_remove(struct platform_device *pdev) @@ -422,9 +415,7 @@ static int snd_mfld_mc_remove(struct platform_device *pdev) struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev); pr_debug("snd_mfld_mc_remove called\n"); - free_irq(platform_get_irq(pdev, 0), mc_drv_ctx); snd_soc_unregister_card(&snd_soc_card_mfld); - kfree(mc_drv_ctx); return 0; } -- cgit v1.2.1 From 5d45ee3cdbca031d3a59422314f4d65dfacdc03b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 10:29:02 +0100 Subject: ASoC: samsung-ac97: Use devm_clk_get() Reviewed-by: Jingoo Han Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index cb88ead98917..1c85999f3fc3 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -440,7 +440,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err1; } - s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + s3c_ac97.ac97_clk = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(s3c_ac97.ac97_clk)) { dev_err(&pdev->dev, "ac97 failed to get ac97_clock\n"); ret = -ENODEV; @@ -480,7 +480,6 @@ err5: err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); err2: iounmap(s3c_ac97.regs); err1: @@ -501,7 +500,6 @@ static int s3c_ac97_remove(struct platform_device *pdev) free_irq(irq_res->start, NULL); clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); iounmap(s3c_ac97.regs); -- cgit v1.2.1 From 25fd0bfd53b96f774f49aa333b02d2f3a07d68fe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 10:34:37 +0100 Subject: ASoC: samsung-ac97: Convert to devm_ioremap_resource() Reviewed-by: Jingoo Han Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 1c85999f3fc3..04d7fd461484 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -417,11 +417,9 @@ static int s3c_ac97_probe(struct platform_device *pdev) return -ENXIO; } - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "ac97")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - return -EBUSY; - } + s3c_ac97.regs = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(s3c_ac97.regs)) + return PTR_ERR(s3c_ac97.regs); s3c_ac97_pcm_out.channel = dmatx_res->start; s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; @@ -433,13 +431,6 @@ static int s3c_ac97_probe(struct platform_device *pdev) init_completion(&s3c_ac97.done); mutex_init(&s3c_ac97.lock); - s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); - if (s3c_ac97.regs == NULL) { - dev_err(&pdev->dev, "Unable to ioremap register region\n"); - ret = -ENXIO; - goto err1; - } - s3c_ac97.ac97_clk = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(s3c_ac97.ac97_clk)) { dev_err(&pdev->dev, "ac97 failed to get ac97_clock\n"); @@ -481,16 +472,13 @@ err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); err2: - iounmap(s3c_ac97.regs); -err1: - release_mem_region(mem_res->start, resource_size(mem_res)); return ret; } static int s3c_ac97_remove(struct platform_device *pdev) { - struct resource *mem_res, *irq_res; + struct resource *irq_res; asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); @@ -501,12 +489,6 @@ static int s3c_ac97_remove(struct platform_device *pdev) clk_disable_unprepare(s3c_ac97.ac97_clk); - iounmap(s3c_ac97.regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); - return 0; } -- cgit v1.2.1 From a16a6c68e8ddf53b51e7b2689ac9cbe999ea8507 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 11:09:55 +0100 Subject: ASoC: psc-ac97: Use devm_ioremap_resource() Acked-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/au1x/psc-ac97.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 8f1862aa7333..f5a392169338 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -383,15 +383,9 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!iores) return -ENODEV; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) -- cgit v1.2.1 From 2105d63e09ebb8273cb09e70b8241c9f89aae544 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 11:28:59 +0100 Subject: ASoC: psc-ac97: Convert to module_platform_driver() Acked-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/au1x/psc-ac97.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index f5a392169338..a97ba1367b69 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -497,19 +497,7 @@ static struct platform_driver au1xpsc_ac97_driver = { .remove = au1xpsc_ac97_drvremove, }; -static int __init au1xpsc_ac97_load(void) -{ - au1xpsc_ac97_workdata = NULL; - return platform_driver_register(&au1xpsc_ac97_driver); -} - -static void __exit au1xpsc_ac97_unload(void) -{ - platform_driver_unregister(&au1xpsc_ac97_driver); -} - -module_init(au1xpsc_ac97_load); -module_exit(au1xpsc_ac97_unload); +module_platform_driver(au1xpsc_ac97_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -- cgit v1.2.1 From d8b51c11ff5a70244753ba60abfd47088cf4dcd4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 11:30:37 +0100 Subject: ASoC: ac97c: Use module_platform_driver() Acked-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 44b8dcecf571..a51dabe20cbb 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -338,19 +338,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -static int __init au1xac97c_load(void) -{ - ac97c_workdata = NULL; - return platform_driver_register(&au1xac97c_driver); -} - -static void __exit au1xac97c_unload(void) -{ - platform_driver_unregister(&au1xac97c_driver); -} - -module_init(au1xac97c_load); -module_exit(au1xac97c_unload); +module_platform_driver(&au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- cgit v1.2.1 From 6dab2fd71cd10756add702edc4b853f3829b8926 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 11:47:56 +0100 Subject: ASoC: bf5xx-ac97: Convert to devm_gpio_request_one() Also clean up the error reporting from failed requests while we're at it. Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 490217325975..024e2dbe6c7f 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -293,13 +293,14 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET /* Request PB3 as reset pin */ - if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) { - pr_err("Failed to request GPIO_%d for reset\n", - CONFIG_SND_BF5XX_RESET_GPIO_NUM); - ret = -1; + ret = devm_gpio_request_one(&pdev->dev, + CONFIG_SND_BF5XX_RESET_GPIO_NUM, + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + dev_err(&pdev->dev, + "Failed to request GPIO_%d for reset: %d\n", + CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); goto gpio_err; } - gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); #endif sport_handle = sport_init(pdev, 2, sizeof(struct ac97_frame), @@ -349,10 +350,6 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) sport_config_err: sport_done(sport_handle); sport_err: -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -gpio_err: -#endif return ret; } @@ -363,9 +360,6 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif return 0; } -- cgit v1.2.1 From 417ced8b93d16f6f90336fdf6929ed599e74f705 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 11:53:45 +0100 Subject: ASoC: ep93xx: Remove redundant dev_set_drvdata() calls The driver core does this and it's never legal to rely on the value of drvdata if not set in probe() anyway. Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 7798fbd5e81d..d49e0556e381 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -405,7 +405,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) fail: platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -420,7 +419,6 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- cgit v1.2.1 From ad3ae47b109358511a807a7859822890c59a06ec Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 12:11:33 +0100 Subject: ASoC: nuc900-ac97: Convert to use devm_ APIs Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 49 +++++++++++------------------------------- 1 file changed, 13 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index fe3285ceaf5b..8dea4c1fd997 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -326,41 +326,32 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) if (nuc900_ac97_data) return -EBUSY; - nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL); + nuc900_audio = devm_kzalloc(&pdev->dev, sizeof(struct nuc900_audio), + GFP_KERNEL); if (!nuc900_audio) return -ENOMEM; spin_lock_init(&nuc900_audio->lock); nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!nuc900_audio->res) { - ret = -ENODEV; - goto out0; - } + if (!nuc900_audio->res) + return ret; - if (!request_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res), pdev->name)) { - ret = -EBUSY; - goto out0; - } - - nuc900_audio->mmio = ioremap(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); - if (!nuc900_audio->mmio) { - ret = -ENOMEM; - goto out1; - } + nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev, + nuc900_audio->res); + if (IS_ERR(nuc900_audio->mmio)) + return PTR_ERR(nuc900_audio->mmio); - nuc900_audio->clk = clk_get(&pdev->dev, NULL); + nuc900_audio->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(nuc900_audio->clk)) { ret = PTR_ERR(nuc900_audio->clk); - goto out2; + goto out; } nuc900_audio->irq_num = platform_get_irq(pdev, 0); if (!nuc900_audio->irq_num) { ret = -EBUSY; - goto out3; + goto out; } nuc900_ac97_data = nuc900_audio; @@ -368,22 +359,14 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) - goto out3; + goto out; /* enbale ac97 multifunction pin */ mfp_set_groupg(nuc900_audio->dev, NULL); return 0; -out3: - clk_put(nuc900_audio->clk); -out2: - iounmap(nuc900_audio->mmio); -out1: - release_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); -out0: - kfree(nuc900_audio); +out: return ret; } @@ -391,12 +374,6 @@ static int nuc900_ac97_drvremove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); - clk_put(nuc900_ac97_data->clk); - iounmap(nuc900_ac97_data->mmio); - release_mem_region(nuc900_ac97_data->res->start, - resource_size(nuc900_ac97_data->res)); - - kfree(nuc900_ac97_data); nuc900_ac97_data = NULL; return 0; -- cgit v1.2.1 From cfe68642cff7c121f38a088a795759fd3500122c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 12:16:50 +0100 Subject: ASoC: tegra20-ac97: Convert to devm_clk_get() Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 2f70ea7f6618..d8c142dc93e4 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -327,7 +327,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ac97); - ac97->clk_ac97 = clk_get(&pdev->dev, NULL); + ac97->clk_ac97 = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ac97->clk_ac97)) { dev_err(&pdev->dev, "Can't retrieve ac97 clock\n"); ret = PTR_ERR(ac97->clk_ac97); @@ -443,7 +443,6 @@ err_unregister_pcm: err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_clk_put: - clk_put(ac97->clk_ac97); err: return ret; } @@ -458,7 +457,6 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&ac97->util_data); clk_disable_unprepare(ac97->clk_ac97); - clk_put(ac97->clk_ac97); return 0; } -- cgit v1.2.1 From 2def2516aad75d73e899ee952754a89774d5b935 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 12:18:39 +0100 Subject: ASoC: tegra20-ac97: Convert to devm_ioremap_resource() Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index d8c142dc93e4..9043626da6fa 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -341,18 +341,10 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_clk_put; } - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + dev_err(&pdev->dev, "ioremap failed: %d\n", ret); goto err_clk_put; } -- cgit v1.2.1 From 65fb3e726c5b21f0c4b76e69d8e1dae961ae74e8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 12:27:58 +0100 Subject: ASoC: tegra-ac97: Do common and clock init prior to component registration Otherwise we may instantiate and hence have something try to access the device while it is still completing initialisation. Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 9043626da6fa..48037f784a86 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -395,23 +395,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->capture_dma_data.maxburst = 4; ac97->capture_dma_data.slave_id = of_dma[0]; - ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, - &tegra20_ac97_dai, 1); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_clk_put; - } - - ret = tegra_pcm_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_component; - } - ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); if (ret) - goto err_unregister_pcm; + goto err_clk_put; ret = tegra_asoc_utils_set_ac97_rate(&ac97->util_data); if (ret) @@ -423,17 +409,31 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_asoc_utils_fini; } + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, + &tegra20_ac97_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_asoc_utils_fini; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_component; + } + /* XXX: crufty ASoC AC97 API - only one AC97 codec allowed */ workdata = ac97; return 0; -err_asoc_utils_fini: - tegra_asoc_utils_fini(&ac97->util_data); err_unregister_pcm: tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_asoc_utils_fini: + tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: return ret; -- cgit v1.2.1 From 3bed3344c82623f6a37f3032e307d9af5b2d7519 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 12:40:47 +0100 Subject: ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource() Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc-ac97.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 8a2840304d28..8ee8d4220014 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -188,9 +188,9 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (!r) return -EBUSY; - if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), - dev_name(&pdev->dev))) - return -EBUSY; + drvdata->base = devm_ioremap_resource(&pdev->dev, r); + if (IS_ERR(drvdata->base)) + return PTR_ERR(drvdata->base); drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); if (!drvdata) @@ -201,9 +201,6 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) r->start >= TXX9_DIRECTMAP_BASE && r->start < TXX9_DIRECTMAP_BASE + 0x400000) drvdata->physbase |= 0xf00000000ull; - drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); - if (!drvdata->base) - return -EBUSY; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, 0, dev_name(&pdev->dev), drvdata); if (err < 0) -- cgit v1.2.1 From 9b86421d14b8780c7abe3c6c8d27e617a74d0148 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 25 Jun 2013 17:19:56 +0100 Subject: ASoC: wm9705: Remove noisy print on boot There's no content in the announcement. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 05b1f346695b..a5fc61dbcd47 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -337,8 +337,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "WM9705 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); -- cgit v1.2.1 From b047e1cce8fe32475ab61846772943a5e4c0a908 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jun 2013 12:45:59 +0100 Subject: ASoC: ac97: Support multi-platform AC'97 Currently we can only have a single platform built in with AC'97 support due to the use of a global variable to provide the bus operations. Fix this by making that variable a pointer and having the bus drivers set the operations prior to registering. This is not a particularly good or nice approach but it avoids blocking multiplatform and a real fix involves fixing the fairly deep problems with AC'97 support - we should be converting it to a real bus. Acked-by: Arnd Bergmann Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 7 +++++-- sound/soc/au1x/psc-ac97.c | 7 +++++-- sound/soc/blackfin/bf5xx-ac97.c | 11 +++++++++-- sound/soc/cirrus/ep93xx-ac97.c | 10 ++++++++-- sound/soc/codecs/ac97.c | 7 ++++--- sound/soc/codecs/ad1980.c | 12 ++++++------ sound/soc/codecs/stac9766.c | 22 +++++++++++----------- sound/soc/codecs/wm9705.c | 14 +++++++------- sound/soc/codecs/wm9712.c | 18 +++++++++--------- sound/soc/codecs/wm9713.c | 18 +++++++++--------- sound/soc/fsl/imx-ssi.c | 11 +++++++++-- sound/soc/fsl/mpc5200_psc_ac97.c | 10 ++++++++-- sound/soc/nuc900/nuc900-ac97.c | 11 ++++++++--- sound/soc/pxa/pxa2xx-ac97.c | 8 ++++++-- sound/soc/samsung/ac97.c | 12 +++++++++--- sound/soc/sh/hac.c | 8 ++++++-- sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 12 ++++++++++-- sound/soc/txx9/txx9aclc-ac97.c | 8 ++++++-- 19 files changed, 151 insertions(+), 71 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index a51dabe20cbb..d6f7694fcad4 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -179,13 +179,12 @@ static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ac97c_bus_ops = { .read = au1xac97c_ac97_read, .write = au1xac97c_ac97_write, .reset = au1xac97c_ac97_cold_reset, .warm_reset = au1xac97c_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */ static int alchemy_ac97c_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -272,6 +271,10 @@ static int au1xac97c_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); + ret = snd_soc_set_ac97_ops(&ac97c_bus_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component, &au1xac97c_dai_driver, 1); if (ret) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a97ba1367b69..a822ab822bb7 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -201,13 +201,12 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = au1xpsc_ac97_read, .write = au1xpsc_ac97_write, .reset = au1xpsc_ac97_cold_reset, .warm_reset = au1xpsc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -417,6 +416,10 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); + ret = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, &wd->dai_drv, 1); if (ret) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 024e2dbe6c7f..c5af677ba49c 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -198,13 +198,12 @@ static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97) #endif } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops bf5xx_ac97_ops = { .read = bf5xx_ac97_read, .write = bf5xx_ac97_write, .warm_reset = bf5xx_ac97_warm_reset, .reset = bf5xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) @@ -336,6 +335,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) goto sport_config_err; } + ret = snd_soc_set_ac97_ops(&bf5xx_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto sport_config_err; + } + ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component, &bfin_ac97_dai, 1); if (ret) { @@ -350,6 +355,7 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) sport_config_err: sport_done(sport_handle); sport_err: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -360,6 +366,7 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index d49e0556e381..4bc9490e2c84 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -237,13 +237,12 @@ static irqreturn_t ep93xx_ac97_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ep93xx_ac97_ops = { .read = ep93xx_ac97_read, .write = ep93xx_ac97_write, .reset = ep93xx_ac97_cold_reset, .warm_reset = ep93xx_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -395,6 +394,10 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); + ret = snd_soc_set_ac97_ops(&ep93xx_ac97_ops); + if (ret) + goto fail; + ret = snd_soc_register_component(&pdev->dev, &ep93xx_ac97_component, &ep93xx_ac97_dai, 1); if (ret) @@ -405,6 +408,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) fail: platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; + snd_soc_set_ac97_ops(NULL); return ret; } @@ -420,6 +424,8 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) platform_set_drvdata(pdev, NULL); ep93xx_ac97_info = NULL; + snd_soc_set_ac97_ops(NULL); + return 0; } diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ef2ae32ffc66..ec7351803c24 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -62,13 +62,13 @@ static struct snd_soc_dai_driver ac97_dai = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); return 0; } @@ -79,7 +79,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) int ret; /* add codec as bus device for standard ac97 */ - ret = snd_ac97_bus(codec->card->snd_card, 0, &soc_ac97_ops, NULL, &ac97_bus); + ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL, + &ac97_bus); if (ret < 0) return ret; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index f385342947d3..89fcf7d6e7b8 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -108,7 +108,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, case AC97_EXTENDED_STATUS: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -124,7 +124,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; @@ -154,13 +154,13 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) u16 retry_cnt = 0; retry: - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, AC97_RESET) == 0x0090) return 1; } - soc_ac97_ops.reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); /* Set bit 16slot in register 74h, then every slot will has only 16 * bits. This command is sent out in 20bit mode, in which case the * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ @@ -186,7 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index cbc7ae322324..a5455c1aea42 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -143,14 +143,14 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); cache[reg / 2] = val; return 0; } @@ -162,7 +162,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } @@ -173,7 +173,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { - val = soc_ac97_ops.read(codec->ac97, reg); + val = soc_ac97_ops->read(codec->ac97, reg); return val; } return cache[reg / 2]; @@ -240,15 +240,15 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; @@ -272,7 +272,7 @@ reset: return -EIO; } codec->ac97->bus->ops->warm_reset(codec->ac97); - id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { stac9766_reset(codec, 0); reset++; @@ -336,7 +336,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index a5fc61dbcd47..70ce6793c5bd 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -209,7 +209,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) case AC97_RESET: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -225,7 +225,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9705_reg))) cache[reg] = val; @@ -294,8 +294,8 @@ static struct snd_soc_dai_driver wm9705_dai[] = { static int wm9705_reset(struct snd_soc_codec *codec) { - if (soc_ac97_ops.reset) { - soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops->reset) { + soc_ac97_ops->reset(codec->ac97); if (ac97_read(codec, 0) == wm9705_reg[0]) return 0; /* Success */ } @@ -306,7 +306,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { - soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); + soc_ac97_ops->write(codec->ac97, AC97_POWERDOWN, 0xffff); return 0; } @@ -323,7 +323,7 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) } for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } return 0; @@ -337,7 +337,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 8e9a6a3eeb1a..c5eb746087b4 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -455,7 +455,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_REC_GAIN) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -472,7 +472,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; @@ -581,15 +581,15 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; @@ -624,7 +624,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || (i > 0x58 && i != 0x5c)) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -635,7 +635,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f7afa68d8c7f..a53e175c015a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -652,7 +652,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_CD) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -668,7 +668,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; @@ -1095,15 +1095,15 @@ static struct snd_soc_dai_driver wm9713_dai[] = { int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; @@ -1180,7 +1180,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || i == AC97_EXTENDED_MSTATUS || i > 0x66) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -1197,7 +1197,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index c6fa03e2114a..bd40849454a8 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -501,13 +501,12 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) imx_ssi_ac97_read(ac97, 0); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops imx_ssi_ac97_ops = { .read = imx_ssi_ac97_read, .write = imx_ssi_ac97_write, .reset = imx_ssi_ac97_reset, .warm_reset = imx_ssi_ac97_warm_reset }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int imx_ssi_probe(struct platform_device *pdev) { @@ -583,6 +582,12 @@ static int imx_ssi_probe(struct platform_device *pdev) platform_set_drvdata(pdev, ssi); + ret = snd_soc_set_ac97_ops(&imx_ssi_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto failed_register; + } + ret = snd_soc_register_component(&pdev->dev, &imx_component, dai, 1); if (ret) { @@ -630,6 +635,7 @@ failed_register: release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -649,6 +655,7 @@ static int imx_ssi_remove(struct platform_device *pdev) release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 4141b35ef0bb..3ef7a0c92efa 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -131,13 +131,12 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97) psc_ac97_warm_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = psc_ac97_read, .write = psc_ac97_write, .reset = psc_ac97_cold_reset, .warm_reset = psc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -290,6 +289,12 @@ static int psc_ac97_of_probe(struct platform_device *op) if (rc != 0) return rc; + rc = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (rc != 0) { + dev_err(&op->dev, "Failed to set AC'97 ops: %d\n", ret); + return rc; + } + rc = snd_soc_register_component(&op->dev, &psc_ac97_component, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); if (rc != 0) { @@ -318,6 +323,7 @@ static int psc_ac97_of_remove(struct platform_device *op) { mpc5200_audio_dma_destroy(op); snd_soc_unregister_component(&op->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 8dea4c1fd997..f4c2417a8730 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -197,13 +197,12 @@ static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops nuc900_ac97_ops = { .read = nuc900_ac97_read, .write = nuc900_ac97_write, .reset = nuc900_ac97_cold_reset, .warm_reset = nuc900_ac97_warm_reset, -} -EXPORT_SYMBOL_GPL(soc_ac97_ops); +}; static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -356,6 +355,10 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) nuc900_ac97_data = nuc900_audio; + ret = snd_soc_set_ac97_ops(&nuc900_ac97_ops); + if (ret) + goto out; + ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) @@ -367,6 +370,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) return 0; out: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -375,6 +379,7 @@ static int nuc900_ac97_drvremove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); nuc900_ac97_data = NULL; + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 57ea8e6c5488..a3c22ba25f08 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -41,13 +41,12 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) pxa2xx_ac97_finish_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .read = pxa2xx_ac97_read, .write = pxa2xx_ac97_write, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { .name = "AC97 PCM Stereo out", @@ -244,6 +243,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return -ENXIO; } + ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); + if (ret != 0) + return ret; + /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -255,6 +258,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 04d7fd461484..2dd623fa3882 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -214,13 +214,12 @@ static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops s3c_ac97_ops = { .read = s3c_ac97_read, .write = s3c_ac97_write, .warm_reset = s3c_ac97_warm_reset, .reset = s3c_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -452,6 +451,12 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err4; } + ret = snd_soc_set_ac97_ops(&s3c_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto err4; + } + ret = snd_soc_register_component(&pdev->dev, &s3c_ac97_component, s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); if (ret) @@ -472,7 +477,7 @@ err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); err2: - + snd_soc_set_ac97_ops(NULL); return ret; } @@ -488,6 +493,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) free_irq(irq_res->start, NULL); clk_disable_unprepare(s3c_ac97.ac97_clk); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index af19f77b7bf0..0af2e4dfd139 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -227,13 +227,12 @@ static void hac_ac97_coldrst(struct snd_ac97 *ac97) hac_ac97_warmrst(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops hac_ac97_ops = { .read = hac_ac97_read, .write = hac_ac97_write, .reset = hac_ac97_coldrst, .warm_reset = hac_ac97_warmrst, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int hac_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -316,6 +315,10 @@ static const struct snd_soc_component_driver sh4_hac_component = { static int hac_soc_platform_probe(struct platform_device *pdev) { + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); + if (ret != 0) + return ret; + return snd_soc_register_component(&pdev->dev, &sh4_hac_component, sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } @@ -323,6 +326,7 @@ static int hac_soc_platform_probe(struct platform_device *pdev) static int hac_soc_platform_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d56bbea6e75e..562d72e04e6e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2079,6 +2079,22 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); +struct snd_ac97_bus_ops *soc_ac97_ops; + +int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) +{ + if (ops == soc_ac97_ops) + return 0; + + if (soc_ac97_ops && ops) + return -EBUSY; + + soc_ac97_ops = ops; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); + /** * snd_soc_free_ac97_codec - free AC97 codec device * @codec: audio codec diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 48037f784a86..f52eab6d2231 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -142,13 +142,12 @@ static void tegra20_ac97_codec_write(struct snd_ac97 *ac97_snd, } while (!time_after(jiffies, timeout)); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops tegra20_ac97_ops = { .read = tegra20_ac97_codec_read, .write = tegra20_ac97_codec_write, .reset = tegra20_ac97_codec_reset, .warm_reset = tegra20_ac97_codec_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) { @@ -409,6 +408,12 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_asoc_utils_fini; } + ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); + if (ret) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto err_asoc_utils_fini; + } + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, &tegra20_ac97_dai, 1); if (ret) { @@ -436,6 +441,7 @@ err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -450,6 +456,8 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) clk_disable_unprepare(ac97->clk_ac97); + snd_soc_set_ac97_ops(NULL); + return 0; } diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 8ee8d4220014..4bcce8a3cded 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -119,12 +119,11 @@ static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops txx9aclc_ac97_ops = { .read = txx9aclc_ac97_read, .write = txx9aclc_ac97_write, .reset = txx9aclc_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) { @@ -206,6 +205,10 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (err < 0) return err; + err = snd_soc_set_ac97_ops(&txx9aclc_ac97_ops); + if (err < 0) + return err; + return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, &txx9aclc_ac97_dai, 1); } @@ -213,6 +216,7 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } -- cgit v1.2.1 From db10e7fbbc836fb66d4500c64c1960940cfad2b0 Mon Sep 17 00:00:00 2001 From: Yijing Wang Date: Thu, 27 Jun 2013 20:55:11 +0800 Subject: ALSA: pci: trivial: replace numeric with standard PM state macros Use standard PM state macros PCI_Dx instead of numeric 0/1/2.. Signed-off-by: Yijing Wang Signed-off-by: Takashi Iwai --- sound/pci/cs4281.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 64659facd155..1dc793e742d7 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1312,7 +1312,7 @@ static int snd_cs4281_free(struct cs4281 *chip) /* Sound System Power Management - Turn Everything OFF */ snd_cs4281_pokeBA0(chip, BA0_SSPM, 0); /* PCI interface - D3 state */ - pci_set_power_state(chip->pci, 3); + pci_set_power_state(chip->pci, PCI_D3hot); if (chip->irq >= 0) free_irq(chip->irq, chip); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 372f8ea91fca..ca8929b9a5d6 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1939,7 +1939,7 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq) #endif if (ensoniq->irq >= 0) synchronize_irq(ensoniq->irq); - pci_set_power_state(ensoniq->pci, 3); + pci_set_power_state(ensoniq->pci, PCI_D3hot); __hw_end: #ifdef CHIP1370 if (ensoniq->dma_bug.area) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 22056c50fe39..d591c154fc58 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2258,7 +2258,7 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) /* FIXME: temporarily disabled, otherwise we cannot fire up * the chip again unless reboot. ACPI bug? */ - pci_set_power_state(chip->pci, 3); + pci_set_power_state(chip->pci, PCI_D3hot); #endif #ifdef CONFIG_PM_SLEEP -- cgit v1.2.1 From 8f898e92aea2c24c7f379ee265d178f69ebb9c07 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 31 Jan 2013 21:39:17 +0100 Subject: ALSA: usb-audio: store protocol version in struct audioformat Instead of reading bInterfaceProtocol from the descriptor whenever it's needed, store this value in the audioformat structure. Besides simplifying some code, this will allow us to correctly handle vendor- specific devices where the descriptors are marked with other values. Signed-off-by: Clemens Ladisch --- sound/usb/card.h | 1 + sound/usb/clock.c | 4 +--- sound/usb/format.c | 34 ++++++++++------------------------ sound/usb/format.h | 2 +- sound/usb/pcm.c | 4 +--- sound/usb/stream.c | 3 ++- 6 files changed, 16 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index bf2889a2cae5..5ecacaa90b53 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -21,6 +21,7 @@ struct audioformat { unsigned char endpoint; /* endpoint */ unsigned char ep_attr; /* endpoint attributes */ unsigned char datainterval; /* log_2 of data packet interval */ + unsigned char protocol; /* UAC_VERSION_1/2 */ unsigned int maxpacksize; /* max. packet size */ unsigned int rates; /* rate bitmasks */ unsigned int rate_min, rate_max; /* min/max rates */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 3a2ce390e278..86f80c60b21f 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -407,9 +407,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate) { - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - - switch (altsd->bInterfaceProtocol) { + switch (fmt->protocol) { case UAC_VERSION_1: default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); diff --git a/sound/usb/format.c b/sound/usb/format.c index 99299ffb33ac..3525231c6b97 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -43,13 +43,12 @@ */ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - unsigned int format, void *_fmt, - int protocol) + unsigned int format, void *_fmt) { int sample_width, sample_bytes; u64 pcm_formats = 0; - switch (protocol) { + switch (fp->protocol) { case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; @@ -360,11 +359,8 @@ err: */ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int format, - struct uac_format_type_i_continuous_descriptor *fmt, - struct usb_host_interface *iface) + struct uac_format_type_i_continuous_descriptor *fmt) { - struct usb_interface_descriptor *altsd = get_iface_desc(iface); - int protocol = altsd->bInterfaceProtocol; snd_pcm_format_t pcm_format; int ret; @@ -387,8 +383,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, } fp->formats = pcm_format_to_bits(pcm_format); } else { - fp->formats = parse_audio_format_i_type(chip, fp, format, - fmt, protocol); + fp->formats = parse_audio_format_i_type(chip, fp, format, fmt); if (!fp->formats) return -EINVAL; } @@ -398,11 +393,8 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, * proprietary class specific descriptor. * audio class v2 uses class specific EP0 range requests for that. */ - switch (protocol) { + switch (fp->protocol) { default: - snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - /* fall through */ case UAC_VERSION_1: fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); @@ -427,12 +419,9 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, */ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *_fmt, - struct usb_host_interface *iface) + int format, void *_fmt) { int brate, framesize, ret; - struct usb_interface_descriptor *altsd = get_iface_desc(iface); - int protocol = altsd->bInterfaceProtocol; switch (format) { case UAC_FORMAT_TYPE_II_AC3: @@ -452,11 +441,8 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, fp->channels = 1; - switch (protocol) { + switch (fp->protocol) { default: - snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - /* fall through */ case UAC_VERSION_1: { struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; brate = le16_to_cpu(fmt->wMaxBitRate); @@ -483,17 +469,17 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int format, struct uac_format_type_i_continuous_descriptor *fmt, - int stream, struct usb_host_interface *iface) + int stream) { int err; switch (fmt->bFormatType) { case UAC_FORMAT_TYPE_I: case UAC_FORMAT_TYPE_III: - err = parse_audio_format_i(chip, fp, format, fmt, iface); + err = parse_audio_format_i(chip, fp, format, fmt); break; case UAC_FORMAT_TYPE_II: - err = parse_audio_format_ii(chip, fp, format, fmt, iface); + err = parse_audio_format_ii(chip, fp, format, fmt); break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", diff --git a/sound/usb/format.h b/sound/usb/format.h index 6f315226f320..4b8a01129f24 100644 --- a/sound/usb/format.h +++ b/sound/usb/format.h @@ -4,6 +4,6 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int format, struct uac_format_type_i_continuous_descriptor *fmt, - int stream, struct usb_host_interface *iface); + int stream); #endif /* __USBAUDIO_FORMAT_H */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 93b6e32cfead..776c58c7cba0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -202,13 +202,11 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt) { - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - /* if endpoint doesn't have pitch control, bail out */ if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; - switch (altsd->bInterfaceProtocol) { + switch (fmt->protocol) { case UAC_VERSION_1: default: return init_pitch_v1(chip, iface, alts, fmt); diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 7db2f8958e79..1ea5871cb980 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -635,6 +635,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; fp->datainterval = snd_usb_parse_datainterval(chip, alts); + fp->protocol = protocol; fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->channels = num_channels; if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) @@ -676,7 +677,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) } /* ok, let's parse further... */ - if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { + if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream) < 0) { kfree(fp->rate_table); kfree(fp->chmap); kfree(fp); -- cgit v1.2.1 From ba7c2be114243fa4cfcbc66a81db18e1d55abf4b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 3 Feb 2013 22:31:20 +0100 Subject: ALSA: usb-audio: detect implicit feedback on Roland devices All the Roland/Edirol/BOSS USB audio devices that need implicit feedback show this unambiguously in their descriptors, so it might be a good idea to let the driver detect this. This should make playback work correctly (at least with Jack) with the following devices: - BOSS GT-100 - BOSS JS-8 Jam Station - Edirol M-16DX - Roland GAIA SH-01 Signed-off-by: Clemens Ladisch --- sound/usb/pcm.c | 41 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 776c58c7cba0..15b151ed4899 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -298,6 +298,35 @@ static int deactivate_endpoints(struct snd_usb_substream *subs) return 0; } +static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, + unsigned int altsetting, + struct usb_host_interface **alts, + unsigned int *ep) +{ + struct usb_interface *iface; + struct usb_interface_descriptor *altsd; + struct usb_endpoint_descriptor *epd; + + iface = usb_ifnum_to_if(dev, ifnum); + if (!iface || iface->num_altsetting < altsetting + 1) + return -ENOENT; + *alts = &iface->altsetting[altsetting]; + altsd = get_iface_desc(*alts); + if (altsd->bAlternateSetting != altsetting || + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC || + (altsd->bInterfaceSubClass != 2 && + altsd->bInterfaceProtocol != 2 ) || + altsd->bNumEndpoints < 1) + return -ENOENT; + epd = get_endpoint(*alts, 0); + if (!usb_endpoint_is_isoc_in(epd) || + (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_IMPLICIT_FB) + return -ENOENT; + *ep = epd->bEndpointAddress; + return 0; +} + /* * find a matching format and set up the interface */ @@ -393,6 +422,18 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) goto add_sync_ep; } } + if (is_playback && + attr == USB_ENDPOINT_SYNC_ASYNC && + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && + altsd->bInterfaceProtocol == 2 && + altsd->bNumEndpoints == 1 && + USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ && + search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1, + altsd->bAlternateSetting, + &alts, &ep) >= 0) { + implicit_fb = 1; + goto add_sync_ep; + } if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && -- cgit v1.2.1 From aafe77cc45a595ca1d4536f2412ddf671ea9108c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 31 Mar 2013 23:43:12 +0200 Subject: ALSA: usb-audio: add support for many Roland/Yamaha devices Add quirks to detect the various vendor-specific descriptors used by Roland and Yamaha in most of their recent USB audio and MIDI devices. Together with the previous patch, this should add audio/MIDI support for the following USB devices: - Edirol motion dive .tokyo performance package - Roland MC-808 Synthesizer - Roland BK-7m Synthesizer - Roland VIMA JM-5/8 Synthesizer - Roland SP-555 Sequencer - Roland V-Synth GT Synthesizer - Roland Music Atelier AT-75/100/300/350C/500/800/900/900C Organ - Edirol V-Mixer M-200i/300/380/400/480/R-1000 - BOSS GT-10B Effects Processor - Roland Fantom G6/G7/G8 Keyboard - Cakewalk Sonar V-Studio 20/100/700 Audio Interface - Roland GW-8 Keyboard - Roland AX-Synth Keyboard - Roland JUNO-Di/STAGE/Gi Keyboard - Roland VB-99 Effects Processor - Cakewalk UM-2G MIDI Interface - Roland A-500S Keyboard - Roland SD-50 Synthesizer - Roland OCTAPAD SPD-30 Controller - Roland Lucina AX-09 Synthesizer - BOSS BR-800 Digital Recorder - Roland DUO/TRI-CAPTURE (EX) Audio Interface - BOSS RC-300 Loop Station - Roland JUPITER-50/80 Keyboard - Roland R-26 Recorder - Roland SPD-SX Controller - BOSS JS-10 Audio Player - Roland TD-11/15/30 Drum Module - Roland A-49/88 Keyboard - Roland INTEGRA-7 Synthesizer - Roland R-88 Recorder Signed-off-by: Clemens Ladisch --- sound/usb/midi.c | 41 +++++++++++ sound/usb/quirks-table.h | 22 ++++++ sound/usb/quirks.c | 175 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/stream.c | 15 +++- sound/usb/usbaudio.h | 2 + 5 files changed, 252 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 8e01fa4991c5..63dd0545a672 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1947,6 +1947,44 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, return snd_usbmidi_detect_endpoints(umidi, endpoint, 1); } +/* + * Detects the endpoints and ports of Roland devices. + */ +static int snd_usbmidi_detect_roland(struct snd_usb_midi* umidi, + struct snd_usb_midi_endpoint_info* endpoint) +{ + struct usb_interface* intf; + struct usb_host_interface *hostif; + u8* cs_desc; + + intf = umidi->iface; + if (!intf) + return -ENOENT; + hostif = intf->altsetting; + /* + * Some devices have a descriptor <06 24 F1 02 >, + * some have standard class descriptors, or both kinds, or neither. + */ + for (cs_desc = hostif->extra; + cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2; + cs_desc += cs_desc[0]) { + if (cs_desc[0] >= 6 && + cs_desc[1] == USB_DT_CS_INTERFACE && + cs_desc[2] == 0xf1 && + cs_desc[3] == 0x02) { + endpoint->in_cables = (1 << cs_desc[4]) - 1; + endpoint->out_cables = (1 << cs_desc[5]) - 1; + return snd_usbmidi_detect_endpoints(umidi, endpoint, 1); + } else if (cs_desc[0] >= 7 && + cs_desc[1] == USB_DT_CS_INTERFACE && + cs_desc[2] == UAC_HEADER) { + return snd_usbmidi_get_ms_info(umidi, endpoint); + } + } + + return -ENODEV; +} + /* * Creates the endpoints and their ports for Midiman devices. */ @@ -2162,6 +2200,9 @@ int snd_usbmidi_create(struct snd_card *card, case QUIRK_MIDI_YAMAHA: err = snd_usbmidi_detect_yamaha(umidi, &endpoints[0]); break; + case QUIRK_MIDI_ROLAND: + err = snd_usbmidi_detect_roland(umidi, &endpoints[0]); + break; case QUIRK_MIDI_MIDIMAN: umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; memcpy(&endpoints[0], quirk->data, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 7f1722f82c89..b47517d47b08 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -455,6 +455,17 @@ YAMAHA_DEVICE(0x7000, "DTX"), YAMAHA_DEVICE(0x7010, "UB99"), #undef YAMAHA_DEVICE #undef YAMAHA_INTERFACE +/* this catches most recent vendor-specific Yamaha devices */ +{ + .match_flags = USB_DEVICE_ID_MATCH_VENDOR | + USB_DEVICE_ID_MATCH_INT_CLASS, + .idVendor = 0x0499, + .bInterfaceClass = USB_CLASS_VENDOR_SPEC, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUTODETECT + } +}, /* * Roland/RolandED/Edirol/BOSS devices @@ -2031,6 +2042,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* this catches most recent vendor-specific Roland devices */ +{ + .match_flags = USB_DEVICE_ID_MATCH_VENDOR | + USB_DEVICE_ID_MATCH_INT_CLASS, + .idVendor = 0x0582, + .bInterfaceClass = USB_CLASS_VENDOR_SPEC, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUTODETECT + } +}, /* Guillemot devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3879eae7e874..5363bcca9494 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include @@ -175,6 +176,178 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return 0; } +static int create_auto_pcm_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver) +{ + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_endpoint_descriptor *epd; + struct uac1_as_header_descriptor *ashd; + struct uac_format_type_i_discrete_descriptor *fmtd; + + /* + * Most Roland/Yamaha audio streaming interfaces have more or less + * standard descriptors, but older devices might lack descriptors, and + * future ones might change, so ensure that we fail silently if the + * interface doesn't look exactly right. + */ + + /* must have a non-zero altsetting for streaming */ + if (iface->num_altsetting < 2) + return -ENODEV; + alts = &iface->altsetting[1]; + altsd = get_iface_desc(alts); + + /* must have an isochronous endpoint for streaming */ + if (altsd->bNumEndpoints < 1) + return -ENODEV; + epd = get_endpoint(alts, 0); + if (!usb_endpoint_xfer_isoc(epd)) + return -ENODEV; + + /* must have format descriptors */ + ashd = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, + UAC_AS_GENERAL); + fmtd = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, + UAC_FORMAT_TYPE); + if (!ashd || ashd->bLength < 7 || + !fmtd || fmtd->bLength < 8) + return -ENODEV; + + return create_standard_audio_quirk(chip, iface, driver, NULL); +} + +static int create_yamaha_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + struct usb_host_interface *alts) +{ + static const struct snd_usb_audio_quirk yamaha_midi_quirk = { + .type = QUIRK_MIDI_YAMAHA + }; + struct usb_midi_in_jack_descriptor *injd; + struct usb_midi_out_jack_descriptor *outjd; + + /* must have some valid jack descriptors */ + injd = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, USB_MS_MIDI_IN_JACK); + outjd = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, USB_MS_MIDI_OUT_JACK); + if (!injd && !outjd) + return -ENODEV; + if (injd && (injd->bLength < 5 || + (injd->bJackType != USB_MS_EMBEDDED && + injd->bJackType != USB_MS_EXTERNAL))) + return -ENODEV; + if (outjd && (outjd->bLength < 6 || + (outjd->bJackType != USB_MS_EMBEDDED && + outjd->bJackType != USB_MS_EXTERNAL))) + return -ENODEV; + return create_any_midi_quirk(chip, iface, driver, &yamaha_midi_quirk); +} + +static int create_roland_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + struct usb_host_interface *alts) +{ + static const struct snd_usb_audio_quirk roland_midi_quirk = { + .type = QUIRK_MIDI_ROLAND + }; + u8 *roland_desc = NULL; + + /* might have a vendor-specific descriptor <06 24 F1 02 ...> */ + for (;;) { + roland_desc = snd_usb_find_csint_desc(alts->extra, + alts->extralen, + roland_desc, 0xf1); + if (!roland_desc) + return -ENODEV; + if (roland_desc[0] < 6 || roland_desc[3] != 2) + continue; + return create_any_midi_quirk(chip, iface, driver, + &roland_midi_quirk); + } +} + +static int create_std_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + struct usb_host_interface *alts) +{ + struct usb_ms_header_descriptor *mshd; + struct usb_ms_endpoint_descriptor *msepd; + + /* must have the MIDIStreaming interface header descriptor*/ + mshd = (struct usb_ms_header_descriptor *)alts->extra; + if (alts->extralen < 7 || + mshd->bLength < 7 || + mshd->bDescriptorType != USB_DT_CS_INTERFACE || + mshd->bDescriptorSubtype != USB_MS_HEADER) + return -ENODEV; + /* must have the MIDIStreaming endpoint descriptor*/ + msepd = (struct usb_ms_endpoint_descriptor *)alts->endpoint[0].extra; + if (alts->endpoint[0].extralen < 4 || + msepd->bLength < 4 || + msepd->bDescriptorType != USB_DT_CS_ENDPOINT || + msepd->bDescriptorSubtype != UAC_MS_GENERAL || + msepd->bNumEmbMIDIJack < 1 || + msepd->bNumEmbMIDIJack > 16) + return -ENODEV; + + return create_any_midi_quirk(chip, iface, driver, NULL); +} + +static int create_auto_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver) +{ + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_endpoint_descriptor *epd; + int err; + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + + /* must have at least one bulk/interrupt endpoint for streaming */ + if (altsd->bNumEndpoints < 1) + return -ENODEV; + epd = get_endpoint(alts, 0); + if (!usb_endpoint_xfer_bulk(epd) || + !usb_endpoint_xfer_int(epd)) + return -ENODEV; + + switch (USB_ID_VENDOR(chip->usb_id)) { + case 0x0499: /* Yamaha */ + err = create_yamaha_midi_quirk(chip, iface, driver, alts); + if (err < 0 && err != -ENODEV) + return err; + break; + case 0x0582: /* Roland */ + err = create_roland_midi_quirk(chip, iface, driver, alts); + if (err < 0 && err != -ENODEV) + return err; + break; + } + + return create_std_midi_quirk(chip, iface, driver, alts); +} + +static int create_autodetect_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + int err; + + err = create_auto_pcm_quirk(chip, iface, driver); + if (err == -ENODEV) + err = create_auto_midi_quirk(chip, iface, driver); + return err; +} + /* * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. @@ -303,9 +476,11 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, static const quirk_func_t quirk_funcs[] = { [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, [QUIRK_COMPOSITE] = create_composite_quirk, + [QUIRK_AUTODETECT] = create_autodetect_quirk, [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, + [QUIRK_MIDI_ROLAND] = create_any_midi_quirk, [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, [QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk, diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 1ea5871cb980..c4339f97226b 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -493,10 +493,10 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) altsd = get_iface_desc(alts); protocol = altsd->bInterfaceProtocol; /* skip invalid one */ - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + if (((altsd->bInterfaceClass != USB_CLASS_AUDIO || + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && + altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC)) && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && - altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || altsd->bNumEndpoints < 1 || le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) continue; @@ -512,6 +512,15 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) continue; + /* + * Roland audio streaming interfaces are marked with protocols + * 0/1/2, but are UAC 1 compatible. + */ + if (USB_ID_VENDOR(chip->usb_id) == 0x0582 && + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && + protocol <= 2) + protocol = UAC_VERSION_1; + chconfig = 0; /* get audio formats */ switch (protocol) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index bc43bcaddf4d..caabe9b3af49 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -72,9 +72,11 @@ struct snd_usb_audio { enum quirk_type { QUIRK_IGNORE_INTERFACE, QUIRK_COMPOSITE, + QUIRK_AUTODETECT, QUIRK_MIDI_STANDARD_INTERFACE, QUIRK_MIDI_FIXED_ENDPOINT, QUIRK_MIDI_YAMAHA, + QUIRK_MIDI_ROLAND, QUIRK_MIDI_MIDIMAN, QUIRK_MIDI_NOVATION, QUIRK_MIDI_RAW_BYTES, -- cgit v1.2.1 From a968782e27f1c5144919edbbaf6f10e8b437ab3e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 9 Feb 2013 10:05:20 +0100 Subject: ALSA: usb-audio: add MIDI port names for some Roland devices Signed-off-by: Clemens Ladisch --- sound/usb/midi.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 63dd0545a672..b901f468b67a 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1575,8 +1575,41 @@ static struct port_info { EXTERNAL_PORT(0x0582, 0x004d, 0, "%s MIDI"), EXTERNAL_PORT(0x0582, 0x004d, 1, "%s 1"), EXTERNAL_PORT(0x0582, 0x004d, 2, "%s 2"), + /* BOSS GT-PRO */ + CONTROL_PORT(0x0582, 0x0089, 0, "%s Control"), /* Edirol UM-3EX */ CONTROL_PORT(0x0582, 0x009a, 3, "%s Control"), + /* Roland VG-99 */ + CONTROL_PORT(0x0582, 0x00b2, 0, "%s Control"), + EXTERNAL_PORT(0x0582, 0x00b2, 1, "%s MIDI"), + /* Cakewalk Sonar V-Studio 100 */ + EXTERNAL_PORT(0x0582, 0x00eb, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x00eb, 1, "%s Control"), + /* Roland VB-99 */ + CONTROL_PORT(0x0582, 0x0102, 0, "%s Control"), + EXTERNAL_PORT(0x0582, 0x0102, 1, "%s MIDI"), + /* Roland A-PRO */ + EXTERNAL_PORT(0x0582, 0x010f, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x010f, 1, "%s 1"), + CONTROL_PORT(0x0582, 0x010f, 2, "%s 2"), + /* Roland SD-50 */ + ROLAND_SYNTH_PORT(0x0582, 0x0114, 0, "%s Synth", 128), + EXTERNAL_PORT(0x0582, 0x0114, 1, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0114, 2, "%s Control"), + /* Roland OCTA-CAPTURE */ + EXTERNAL_PORT(0x0582, 0x0120, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0120, 1, "%s Control"), + EXTERNAL_PORT(0x0582, 0x0121, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0121, 1, "%s Control"), + /* Roland SPD-SX */ + CONTROL_PORT(0x0582, 0x0145, 0, "%s Control"), + EXTERNAL_PORT(0x0582, 0x0145, 1, "%s MIDI"), + /* Roland A-Series */ + CONTROL_PORT(0x0582, 0x0156, 0, "%s Keyboard"), + EXTERNAL_PORT(0x0582, 0x0156, 1, "%s MIDI"), + /* Roland INTEGRA-7 */ + ROLAND_SYNTH_PORT(0x0582, 0x015b, 0, "%s Synth", 128), + CONTROL_PORT(0x0582, 0x015b, 1, "%s Control"), /* M-Audio MidiSport 8x8 */ CONTROL_PORT(0x0763, 0x1031, 8, "%s Control"), CONTROL_PORT(0x0763, 0x1033, 8, "%s Control"), -- cgit v1.2.1 From 8e5ced83dd1c3090c96c4e0614703f0f2a5ba2f4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 7 Feb 2013 22:45:16 +0100 Subject: ALSA: usb-audio: remove superfluous Roland quirks Remove all quirks that are no longer needed now that the generic Roland quirks can handle the vendor-specific descriptors correctly. Signed-off-by: Clemens Ladisch --- sound/usb/quirks-table.h | 471 ----------------------------------------------- 1 file changed, 471 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b47517d47b08..d8822deaba69 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1141,7 +1141,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Roland M-1000 support */ { /* * Has ID 0x0038 when not in "Advanced Driver" mode; @@ -1256,7 +1255,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Edirol M-100FX support */ { /* has ID 0x004e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x004c), @@ -1375,20 +1373,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -{ - /* has ID 0x006b when not in "Advanced Driver" mode */ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x006a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "SP-606", - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - } -}, { /* has ID 0x006e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x006d), @@ -1476,8 +1460,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Roland V-SYNTH XT support */ - /* TODO: add BOSS GT-PRO support */ { /* has ID 0x008c when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x008b), @@ -1491,42 +1473,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .in_cables = 0x0001 } } -}, - /* TODO: add Edirol PC-80 support */ -{ - USB_DEVICE(0x0582, 0x0096), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "UA-1EX", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x009a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "UM-3EX", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x000f, - .in_cables = 0x000f - } - } }, { /* @@ -1557,125 +1503,9 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } } -}, - /* TODO: add Edirol MD-P1 support */ -{ - USB_DEVICE(0x582, 0x00a6), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "Juno-G", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - } -}, -{ - /* Roland SH-201 */ - USB_DEVICE(0x0582, 0x00ad), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "SH-201", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 - * kHz. In standard mode, the device has ID 0582:00b3, and offers - * 16-bit PCM at 44.1 kHz with no MIDI. - */ - USB_DEVICE(0x0582, 0x00b2), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "VG-99", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - /* Roland SonicCell */ - USB_DEVICE(0x0582, 0x00c2), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "SonicCell", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } }, { /* Edirol M-16DX */ - /* FIXME: This quirk gives a good-working capture stream but the - * playback seems problematic because of lacking of sync - * with capture stream. It needs to sync with the capture - * clock. As now, you'll get frequent sound distortions - * via the playback. - */ USB_DEVICE(0x0582, 0x00c4), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, @@ -1703,35 +1533,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -{ - /* BOSS GT-10 */ - USB_DEVICE(0x0582, 0x00da), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, { /* Advanced modes of the Edirol UA-25EX. * For the standard mode, UA-25EX has ID 0582:00e7, which @@ -1762,42 +1563,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -{ - /* has ID 0x00ea when not in Advanced Driver mode */ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e9), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "Roland", */ - /* .product_name = "UA-1G", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x0104), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "Roland", */ - /* .product_name = "UM-1G", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - } -}, { /* Edirol UM-3G */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), @@ -1806,242 +1571,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_STANDARD_INTERFACE } }, -{ - /* Boss JS-8 Jam Station */ - USB_DEVICE(0x0582, 0x0109), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "JS-8", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, -{ - /* has ID 0x0110 when not in Advanced Driver mode */ - USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "Roland", */ - /* .product_name = "A-PRO", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0007 - } - } -}, -{ - /* Roland GAIA SH-01 */ - USB_DEVICE(0x0582, 0x0111), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "GAIA", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &(const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x0113), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "ME-25", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x0127), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "Roland", */ - /* .product_name = "GR-55", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, -{ - /* Added support for Roland UM-ONE which differs from UM-1 */ - USB_DEVICE(0x0582, 0x012a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "ROLAND", */ - /* .product_name = "UM-ONE", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0003 - } - } -}, -{ - USB_DEVICE(0x0582, 0x011e), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "BR-800", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x0130), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "MICRO BR-80", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ - USB_DEVICE(0x0582, 0x014d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - /* .vendor_name = "BOSS", */ - /* .product_name = "GT-100", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | -- cgit v1.2.1 From b1ce7ba619d9de53db7fad25f445ca9abc2b63df Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 4 Apr 2013 21:43:57 +0200 Subject: ALSA: usb-audio: claim autodetected PCM interfaces all at once snd_card_register() registers all devices newly added since the last call. However, the playback/capture streams are handled as one ALSA device, so the second /dev device will not be registered if the PCM streams are added in two steps. QUIRK_AUTODETECT caused the probe callback to be called once for each interface, which triggered this problem. Work around this by handling this like the composite quirk, i.e., autodetecting all other interfaces that might be used for PCM or MIDI. Signed-off-by: Clemens Ladisch --- sound/usb/quirks.c | 40 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 37 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5363bcca9494..5b01330b8452 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -337,8 +337,7 @@ static int create_auto_midi_quirk(struct snd_usb_audio *chip, static int create_autodetect_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, - struct usb_driver *driver, - const struct snd_usb_audio_quirk *quirk) + struct usb_driver *driver) { int err; @@ -348,6 +347,41 @@ static int create_autodetect_quirk(struct snd_usb_audio *chip, return err; } +static int create_autodetect_quirks(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + int probed_ifnum = get_iface_desc(iface->altsetting)->bInterfaceNumber; + int ifcount, ifnum, err; + + err = create_autodetect_quirk(chip, iface, driver); + if (err < 0) + return err; + + /* + * ALSA PCM playback/capture devices cannot be registered in two steps, + * so we have to claim the other corresponding interface here. + */ + ifcount = chip->dev->actconfig->desc.bNumInterfaces; + for (ifnum = 0; ifnum < ifcount; ifnum++) { + if (ifnum == probed_ifnum || quirk->ifnum >= 0) + continue; + iface = usb_ifnum_to_if(chip->dev, ifnum); + if (!iface || + usb_interface_claimed(iface) || + get_iface_desc(iface->altsetting)->bInterfaceClass != + USB_CLASS_VENDOR_SPEC) + continue; + + err = create_autodetect_quirk(chip, iface, driver); + if (err >= 0) + usb_driver_claim_interface(driver, iface, (void *)-1L); + } + + return 0; +} + /* * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. @@ -476,7 +510,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, static const quirk_func_t quirk_funcs[] = { [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, [QUIRK_COMPOSITE] = create_composite_quirk, - [QUIRK_AUTODETECT] = create_autodetect_quirk, + [QUIRK_AUTODETECT] = create_autodetect_quirks, [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, -- cgit v1.2.1 From b7f33917bcd993ff81f3f80b9dc1890fb7410c6d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 16 Jun 2013 18:27:56 +0200 Subject: ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE The Roland Quad/Octo-Capture devices use some unknown vendor-specific mechanism to switch sample rates (and to manage other controls). To prevent the driver from attempting to use any other than the default 44.1 kHz sample rate, use quirks to hide the other alternate settings. Signed-off-by: Clemens Ladisch --- sound/usb/quirks-table.h | 134 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 134 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d8822deaba69..4ce96b4ddd31 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1571,6 +1571,140 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_STANDARD_INTERFACE } }, +{ + /* only 44.1 kHz works at the moment */ + USB_DEVICE(0x0582, 0x0120), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "OCTO-CAPTURE", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 12, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x85, + .ep_attr = 0x25, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, +{ + /* only 44.1 kHz works at the moment */ + USB_DEVICE(0x0582, 0x012f), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "QUAD-CAPTURE", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 6, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x85, + .ep_attr = 0x25, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | -- cgit v1.2.1 From 6c50e9147ff03996417e24a11e31831d245b52f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 28 Jun 2013 10:57:26 +0100 Subject: ASoC: mfld: Remove unused variable Reported-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index aec29a805354..ee363845759e 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -412,8 +412,6 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) static int snd_mfld_mc_remove(struct platform_device *pdev) { - struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); snd_soc_unregister_card(&snd_soc_card_mfld); return 0; -- cgit v1.2.1 From ad60d502fb8aaa3c1e011f4d72b8228f553d87a8 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 28 Jun 2013 12:03:01 +0200 Subject: ALSA: hda - Add support for ALC5505 DSP power-save mode This patch adds the power-saving control for ALC5505 DSP on some Realtek codecs. Signed-off-by: Kailang Yang Tested-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 98 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 98 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ae121113f223..eeb6ecc31366 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -115,6 +115,7 @@ struct alc_spec { int init_amp; int codec_variant; /* flag for other variants */ + bool has_alc5505_dsp; /* for PLL fix */ hda_nid_t pll_nid; @@ -2580,7 +2581,96 @@ static void alc269_shutup(struct hda_codec *codec) } } +static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, + unsigned int val) +{ + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_COEF_INDEX, index_reg >> 1); + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_PROC_COEF, val & 0xffff); /* LSB */ + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_PROC_COEF, val >> 16); /* MSB */ +} + +static int alc5505_coef_get(struct hda_codec *codec, unsigned int index_reg) +{ + unsigned int val; + + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_COEF_INDEX, index_reg >> 1); + val = snd_hda_codec_read(codec, 0x51, 0, AC_VERB_GET_PROC_COEF, 0) + & 0xffff; + val |= snd_hda_codec_read(codec, 0x51, 0, AC_VERB_GET_PROC_COEF, 0) + << 16; + return val; +} + +static void alc5505_dsp_halt(struct hda_codec *codec) +{ + unsigned int val; + + alc5505_coef_set(codec, 0x3000, 0x000c); /* DSP CPU stop */ + alc5505_coef_set(codec, 0x880c, 0x0008); /* DDR enter self refresh */ + alc5505_coef_set(codec, 0x61c0, 0x11110080); /* Clock control for PLL and CPU */ + alc5505_coef_set(codec, 0x6230, 0xfc0d4011); /* Disable Input OP */ + alc5505_coef_set(codec, 0x61b4, 0x040a2b03); /* Stop PLL2 */ + alc5505_coef_set(codec, 0x61b0, 0x00005b17); /* Stop PLL1 */ + alc5505_coef_set(codec, 0x61b8, 0x04133303); /* Stop PLL3 */ + val = alc5505_coef_get(codec, 0x6220); + alc5505_coef_set(codec, 0x6220, (val | 0x3000)); /* switch Ringbuffer clock to DBUS clock */ +} + +static void alc5505_dsp_back_from_halt(struct hda_codec *codec) +{ + alc5505_coef_set(codec, 0x61b8, 0x04133302); + alc5505_coef_set(codec, 0x61b0, 0x00005b16); + alc5505_coef_set(codec, 0x61b4, 0x040a2b02); + alc5505_coef_set(codec, 0x6230, 0xf80d4011); + alc5505_coef_set(codec, 0x6220, 0x2002010f); + alc5505_coef_set(codec, 0x880c, 0x00000004); +} + +static void alc5505_dsp_init(struct hda_codec *codec) +{ + unsigned int val; + + alc5505_dsp_halt(codec); + alc5505_dsp_back_from_halt(codec); + alc5505_coef_set(codec, 0x61b0, 0x5b14); /* PLL1 control */ + alc5505_coef_set(codec, 0x61b0, 0x5b16); + alc5505_coef_set(codec, 0x61b4, 0x04132b00); /* PLL2 control */ + alc5505_coef_set(codec, 0x61b4, 0x04132b02); + alc5505_coef_set(codec, 0x61b8, 0x041f3300); /* PLL3 control*/ + alc5505_coef_set(codec, 0x61b8, 0x041f3302); + snd_hda_codec_write(codec, 0x51, 0, AC_VERB_SET_CODEC_RESET, 0); /* Function reset */ + alc5505_coef_set(codec, 0x61b8, 0x041b3302); + alc5505_coef_set(codec, 0x61b8, 0x04173302); + alc5505_coef_set(codec, 0x61b8, 0x04163302); + alc5505_coef_set(codec, 0x8800, 0x348b328b); /* DRAM control */ + alc5505_coef_set(codec, 0x8808, 0x00020022); /* DRAM control */ + alc5505_coef_set(codec, 0x8818, 0x00000400); /* DRAM control */ + + val = alc5505_coef_get(codec, 0x6200) >> 16; /* Read revision ID */ + if (val <= 3) + alc5505_coef_set(codec, 0x6220, 0x2002010f); /* I/O PAD Configuration */ + else + alc5505_coef_set(codec, 0x6220, 0x6002018f); + + alc5505_coef_set(codec, 0x61ac, 0x055525f0); /**/ + alc5505_coef_set(codec, 0x61c0, 0x12230080); /* Clock control */ + alc5505_coef_set(codec, 0x61b4, 0x040e2b02); /* PLL2 control */ + alc5505_coef_set(codec, 0x61bc, 0x010234f8); /* OSC Control */ + alc5505_coef_set(codec, 0x880c, 0x00000004); /* DRAM Function control */ + alc5505_coef_set(codec, 0x880c, 0x00000003); + alc5505_coef_set(codec, 0x880c, 0x00000010); +} + #ifdef CONFIG_PM +static int alc269_suspend(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec->has_alc5505_dsp) + alc5505_dsp_halt(codec); + return alc_suspend(codec); +} + static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2605,6 +2695,8 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); + if (spec->has_alc5505_dsp) + alc5505_dsp_back_from_halt(codec); return 0; } #endif /* CONFIG_PM */ @@ -3730,6 +3822,11 @@ static int patch_alc269(struct hda_codec *codec) break; } + if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { + spec->has_alc5505_dsp = true; + spec->init_hook = alc5505_dsp_init; + } + /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) @@ -3740,6 +3837,7 @@ static int patch_alc269(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM + codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; #endif spec->shutup = alc269_shutup; -- cgit v1.2.1 From 6c29d68a82ec68db18241b818d03e7864c052be9 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 28 Jun 2013 08:53:34 +0200 Subject: ALSA: hda - Yet another Dell headset mic quirk This quirk is needed for the headset mic to work on this Dell machine. BugLink: https://bugs.launchpad.net/bugs/1195597 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eeb6ecc31366..065718f0f702 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3601,6 +3601,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- cgit v1.2.1 From cd6fb6793a33e2b02af6e05a8d3f735f7c88a943 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 28 Jun 2013 11:09:56 +0200 Subject: ALSA: hda - Guess what, it's two more Dell headset mic quirks Add two more machines that need quirks for headset mics to work. Tested-by: Shawn Wang BugLink: https://bugs.launchpad.net/bugs/1195636 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 065718f0f702..7d6a9f5d2b06 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3598,6 +3598,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- cgit v1.2.1 From 0c055b3413868227f2e85701c4e6938c9581f0e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Jun 2013 11:51:32 +0200 Subject: ALSA: hda - Fix the max length of control name in generic parser add_control_with_pfx() in hda_generic.c assumes a shorter name string for the control element, and this resulted in the truncation of the long but valid string like "Headphone Surround Switch" in the middle. This patch aligns the max size to the actual limit of snd_ctl_elem_id, 44. Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1485d871d628..6460fc519d36 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -854,7 +854,7 @@ static int add_control_with_pfx(struct hda_gen_spec *spec, int type, const char *pfx, const char *dir, const char *sfx, int cidx, unsigned long val) { - char name[32]; + char name[44]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); if (!add_control(spec, type, name, cidx, val)) return -ENOMEM; -- cgit v1.2.1 From 975cc02a904ae385721f1bdb65eb1bcf707dfaf1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Jun 2013 11:56:49 +0200 Subject: ALSA: Replace the magic number 44 with const The char arrays with size 44 are for the name string of snd_ctl_elem_id. Define the constant and replace the raw numbers with it for clarifying better. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 2 +- sound/pci/asihpi/asihpi.c | 2 +- sound/pci/hda/hda_generic.c | 12 ++++++------ sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/patch_ca0132.c | 8 ++++---- 5 files changed, 13 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index d37c683cfd7a..445ca481d8d3 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1296,7 +1296,7 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, struct snd_ac97 *ac97) { int err; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; unsigned char lo_max, hi_max; if (! snd_ac97_valid_reg(ac97, reg)) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index fbc17203613c..185d54a5cb1a 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1278,7 +1278,7 @@ struct hpi_control { u16 dst_node_type; u16 dst_node_index; u16 band; - char name[44]; /* copied to snd_ctl_elem_id.name[44]; */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* copied to snd_ctl_elem_id.name[44]; */ }; static const char * const asihpi_tuner_band_names[] = { diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 6460fc519d36..8e77cbbad871 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -854,7 +854,7 @@ static int add_control_with_pfx(struct hda_gen_spec *spec, int type, const char *pfx, const char *dir, const char *sfx, int cidx, unsigned long val) { - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); if (!add_control(spec, type, name, cidx, val)) return -ENOMEM; @@ -1931,7 +1931,7 @@ static int create_extra_outs(struct hda_codec *codec, int num_pins, for (i = 0; i < num_pins; i++) { const char *name; - char tmp[44]; + char tmp[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int err, idx = 0; if (num_pins == 2 && i == 1 && !strcmp(pfx, "Speaker")) @@ -2484,7 +2484,7 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins, } if (get_out_jack_num_items(codec, pin) > 1) { struct snd_kcontrol_new *knew; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; get_jack_mode_name(codec, pin, name, sizeof(name)); knew = snd_hda_gen_add_kctl(spec, name, &out_jack_mode_enum); @@ -2616,7 +2616,7 @@ static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin) { struct hda_gen_spec *spec = codec->spec; struct snd_kcontrol_new *knew; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; unsigned int defcfg; if (pin == spec->hp_mic_pin) @@ -3316,7 +3316,7 @@ static int add_single_cap_ctl(struct hda_codec *codec, const char *label, bool inv_dmic) { struct hda_gen_spec *spec = codec->spec; - char tmpname[44]; + char tmpname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = is_switch ? HDA_CTL_WIDGET_MUTE : HDA_CTL_WIDGET_VOL; const char *sfx = is_switch ? "Switch" : "Volume"; unsigned int chs = inv_dmic ? 1 : 3; @@ -3578,7 +3578,7 @@ static int parse_mic_boost(struct hda_codec *codec) struct nid_path *path; unsigned int val; int idx; - char boost_label[44]; + char boost_label[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; idx = imux->items[i].index; if (idx >= imux->num_items) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 9e0a95288f46..3fd2973183e2 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -398,7 +398,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, const char *base_name) { unsigned int def_conf, conn; - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int idx, err; bool phantom_jack; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 90ff7a3f72df..6e9876f27d95 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -139,7 +139,7 @@ enum { #define DSP_SPEAKER_OUT_LATENCY 7 struct ct_effect { - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; hda_nid_t nid; int mid; /*effect module ID*/ int reqs[EFFECT_VALS_MAX_COUNT]; /*effect module request*/ @@ -270,7 +270,7 @@ enum { }; struct ct_tuning_ctl { - char name[44]; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; hda_nid_t parent_nid; hda_nid_t nid; int mid; /*effect module ID*/ @@ -3103,7 +3103,7 @@ static int add_tuning_control(struct hda_codec *codec, hda_nid_t pnid, hda_nid_t nid, const char *name, int dir) { - char namestr[44]; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type); @@ -3935,7 +3935,7 @@ static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag, static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int dir) { - char namestr[44]; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type); -- cgit v1.2.1 From 9f24dc877093744c0db323cc3d8a9c82aa2af8a5 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 27 Jun 2013 21:59:10 +0200 Subject: ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size The TAS5086_CLOCK_CONTROL also has a size of 1 byte. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 72067f79633e..8bbdf25530ca 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -130,7 +130,7 @@ static const struct reg_default tas5086_reg_defaults[] = { static int tas5086_register_size(struct device *dev, unsigned int reg) { switch (reg) { - case TAS5086_DEV_ID ... TAS5086_BKNDERR: + case TAS5086_CLOCK_CONTROL ... TAS5086_BKNDERR: return 1; case TAS5086_INPUT_MUX: case TAS5086_PWM_OUTPUT_MUX: -- cgit v1.2.1 From a975873a9acc0788c1aee5ca183deb420b5c00e5 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 27 Jun 2013 21:59:11 +0200 Subject: ASoC: tas5086: fix Mid-Z implementation It turns out that the TAS5086 doesn't like channel start parts to be empty, and if all channels are configured to Mid-Z, part 1 has to be used. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 8bbdf25530ca..6d31d88f7204 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -721,7 +721,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int charge_period = 1300000; /* hardware default is 1300 ms */ - u8 pwm_start = TAS5086_PWM_START_CHANNEL_MASK; + u8 pwm_start_mid_z = 0; int i, ret; if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { @@ -735,16 +735,19 @@ static int tas5086_probe(struct snd_soc_codec *codec) "ti,mid-z-channel-%d", i + 1); if (of_get_property(of_node, name, NULL) != NULL) - pwm_start &= ~(1 << i); + pwm_start_mid_z |= 1 << i; } } /* - * Configure 'part 2' of the PWM starts to always use MID-Z, and tell - * all configured mid-z channels to start start under 'part 2'. + * If any of the channels is configured to start in Mid-Z mode, + * configure 'part 1' of the PWM starts to use Mid-Z, and tell + * all configured mid-z channels to start start under 'part 1'. */ - regmap_write(priv->regmap, TAS5086_PWM_START, - TAS5086_PWM_START_MIDZ_FOR_START_2 | pwm_start); + if (pwm_start_mid_z) + regmap_write(priv->regmap, TAS5086_PWM_START, + TAS5086_PWM_START_MIDZ_FOR_START_1 | + pwm_start_mid_z); /* lookup and set split-capacitor charge period */ if (charge_period == 0) { -- cgit v1.2.1 From 066624c6a1733a72a67f1d06d35a2153e7d9082b Mon Sep 17 00:00:00 2001 From: Przemek Rudy Date: Thu, 27 Jun 2013 23:52:33 +0200 Subject: ALSA: usb-audio: Add Audio Advantage Micro II This patch is adding extensive support (beside standard usb audio class) for Audio Advantage Micro II usb sound card. Features included: - Access to AES bits (so now sending the IEC61937 compliant stream is possible). - Mixer SPDIF control added to turn on/off the optical transmitter. Signed-off-by: Przemek Rudy Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 212 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 12 +++ 2 files changed, 224 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe91440a068..d42a584cf829 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -9,6 +9,8 @@ * Alan Cox (alan@lxorguk.ukuu.org.uk) * Thomas Sailer (sailer@ife.ee.ethz.ch) * + * Audio Advantage Micro II support added by: + * Przemek Rudy (prudy1@o2.pl) * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -30,6 +32,7 @@ #include #include +#include #include #include #include @@ -1315,6 +1318,211 @@ static struct std_mono_table ebox44_table[] = { {} }; +/* Audio Advantage Micro II findings: + * + * Mapping spdif AES bits to vendor register.bit: + * AES0: [0 0 0 0 2.3 2.2 2.1 2.0] - default 0x00 + * AES1: [3.3 3.2.3.1.3.0 2.7 2.6 2.5 2.4] - default: 0x01 + * AES2: [0 0 0 0 0 0 0 0] + * AES3: [0 0 0 0 0 0 x 0] - 'x' bit is set basing on standard usb request + * (UAC_EP_CS_ATTR_SAMPLE_RATE) for Audio Devices + * + * power on values: + * r2: 0x10 + * r3: 0x20 (b7 is zeroed just before playback (except IEC61937) and set + * just after it to 0xa0, presumably it disables/mutes some analog + * parts when there is no audio.) + * r9: 0x28 + * + * Optical transmitter on/off: + * vendor register.bit: 9.1 + * 0 - on (0x28 register value) + * 1 - off (0x2a register value) + * + */ +static int snd_microii_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned int ep; + unsigned char data[3]; + int rate; + + ucontrol->value.iec958.status[0] = kcontrol->private_value & 0xff; + ucontrol->value.iec958.status[1] = (kcontrol->private_value >> 8) & 0xff; + ucontrol->value.iec958.status[2] = 0x00; + + /* use known values for that card: interface#1 altsetting#1 */ + iface = usb_ifnum_to_if(mixer->chip->dev, 1); + alts = &iface->altsetting[1]; + ep = get_endpoint(alts, 0)->bEndpointAddress; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_rcvctrlpipe(mixer->chip->dev, 0), + UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + ep, + data, + sizeof(data)); + if (err < 0) + goto end; + + rate = data[0] | (data[1] << 8) | (data[2] << 16); + ucontrol->value.iec958.status[3] = (rate == 48000) ? + IEC958_AES3_CON_FS_48000 : IEC958_AES3_CON_FS_44100; + + err = 0; +end: + return err; +} + +static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg; + unsigned long priv_backup = kcontrol->private_value; + + reg = ((ucontrol->value.iec958.status[1] & 0x0f) << 4) | + (ucontrol->value.iec958.status[0] & 0x0f); + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 2, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xfffff0f0; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0x0f) << 8; + kcontrol->private_value |= (ucontrol->value.iec958.status[0] & 0x0f); + + reg = (ucontrol->value.iec958.status[0] & IEC958_AES0_NONAUDIO) ? + 0xa0 : 0x20; + reg |= (ucontrol->value.iec958.status[1] >> 4) & 0x0f; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 3, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xffff0fff; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0xf0) << 8; + + /* The frequency bits in AES3 cannot be set via register access. */ + + /* Silently ignore any bits from the request that cannot be set. */ + + err = (priv_backup != kcontrol->private_value); +end: + return err; +} + +static int snd_microii_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.iec958.status[0] = 0x0f; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0x00; + ucontrol->value.iec958.status[3] = 0x00; + + return 0; +} + +static int snd_microii_spdif_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = !(kcontrol->private_value & 0x02); + + return 0; +} + +static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg = ucontrol->value.integer.value[0] ? 0x28 : 0x2a; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 9, + NULL, + 0); + + if (!err) { + err = (reg != (kcontrol->private_value & 0x0ff)); + if (err) + kcontrol->private_value = reg; + } + + return err; +} + +static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_default_get, + .put = snd_microii_spdif_default_put, + .private_value = 0x00000100UL,/* reset value */ + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_mask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = snd_microii_spdif_switch_get, + .put = snd_microii_spdif_switch_put, + .private_value = 0x00000028UL,/* reset value */ + } +}; + +static int snd_microii_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_microii_mixer_spdif); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_microii_mixer_spdif[i], mixer)); + if (err < 0) + return err; + } + + return err; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1353,6 +1561,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_xonar_u1_controls_create(mixer); break; + case USB_ID(0x0d8c, 0x0103): /* Audio Advantage Micro II */ + err = snd_microii_controls_create(mixer); + break; + case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9c636c2d99f6..f5f0595ef9c7 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3119,4 +3119,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ + USB_DEVICE(0x0d8c, 0x0103), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .product_name = "Audio Advantage MicroII", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- cgit v1.2.1 From 128521f6017d36d7c01449d4c97b37dc10c93387 Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Fri, 28 Jun 2013 21:53:24 +0200 Subject: ASoC: tegra20-ac97: Remove duplicate error message devm_ioremap_resource() already outputs an error message when any of the operations it performs fails, so the duplicate in the caller can be removed. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index f52eab6d2231..6c1255b44535 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -343,7 +343,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) regs = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(regs)) { ret = PTR_ERR(regs); - dev_err(&pdev->dev, "ioremap failed: %d\n", ret); goto err_clk_put; } -- cgit v1.2.1 From 8a08f4c4f24b4dfe7ff08542868e2b434c96221f Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Fri, 28 Jun 2013 21:53:25 +0200 Subject: ASoC: tegra20-ac97: Remove unused variable With the conversion to devm_ioremap_resource() the memregion variable is no longer used so it can be dropped. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 6c1255b44535..e58233f7df61 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -312,7 +312,7 @@ static const struct regmap_config tegra20_ac97_regmap_config = { static int tegra20_ac97_platform_probe(struct platform_device *pdev) { struct tegra20_ac97 *ac97; - struct resource *mem, *memregion; + struct resource *mem; u32 of_dma[2]; void __iomem *regs; int ret = 0; -- cgit v1.2.1 From 7685e0165b36ae034dd5e67b7fbbee7e74604f38 Mon Sep 17 00:00:00 2001 From: Kevin Hilman Date: Fri, 28 Jun 2013 11:17:48 -0700 Subject: ASoC: pxa2xx: fixup multi-platform AC'97 build failures commit b047e1cc (ASoC: ac97: Support multi-platform AC'97) introduced some build failures for the pxa2xx-ac97 support, fix them. Cc: Mark Brown Signed-off-by: Kevin Hilman Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-ac97.c | 2 ++ sound/soc/pxa/pxa2xx-ac97.h | 3 --- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index a3c22ba25f08..1475515712e6 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -238,6 +238,8 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h index eda891e6f31b..a49c21ba3842 100644 --- a/sound/soc/pxa/pxa2xx-ac97.h +++ b/sound/soc/pxa/pxa2xx-ac97.h @@ -14,7 +14,4 @@ #define PXA2XX_DAI_AC97_AUX 1 #define PXA2XX_DAI_AC97_MIC 2 -/* platform data */ -extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; - #endif -- cgit v1.2.1 From f74b5e253a062004c1d30177f9889501423e403d Mon Sep 17 00:00:00 2001 From: Kevin Hilman Date: Fri, 28 Jun 2013 11:17:49 -0700 Subject: ASoC: ac97: fixup multi-platform AC'97 module build failure commit b047e1cc (ASoC: ac97: Support multi-platform AC'97) introduced some build failures for modules wanting to access the generic soc_ac97_ops. For example: ERROR: "soc_ac97_ops" [sound/soc/codecs/snd-soc-wm9712.ko] undefined! To fix, export soc_ac97_ops to modules. Cc: Mark Brown Signed-off-by: Kevin Hilman Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 562d72e04e6e..0a8a5f50b466 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2080,6 +2080,7 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); struct snd_ac97_bus_ops *soc_ac97_ops; +EXPORT_SYMBOL_GPL(soc_ac97_ops); int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) { -- cgit v1.2.1 From ee441140e7676766b0ce8b9e9a259066bb54c149 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 27 Jun 2013 22:00:04 +0200 Subject: ASoC: adau1701: more direct regmap usage Replace calls to snd_soc_update_bits() with regmap_update_bits(). Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 0e250f118c0e..4cd4dd10fa24 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -334,7 +334,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; } - snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, mask, val); return 0; } @@ -362,7 +362,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_SERICTL, + regmap_update_bits(adau1701->regmap, ADAU1701_SERICTL, ADAU1701_SERICTL_MODE_MASK, val); return 0; @@ -403,7 +403,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val); format = params_format(params); @@ -490,6 +490,7 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: @@ -498,11 +499,13 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Enable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, 0x00); break; case SND_SOC_BIAS_OFF: /* Disable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, mask); break; } @@ -514,6 +517,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; unsigned int mask = ADAU1701_DSPCTRL_DAM; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int val; if (mute) @@ -521,7 +525,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) else val = mask; - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, mask, val); return 0; } @@ -543,7 +547,8 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + regmap_update_bits(adau1701->regmap, ADAU1701_OSCIPOW, + ADAU1701_OSCIPOW_OPD, val); adau1701->sysclk = freq; return 0; -- cgit v1.2.1 From cef929ec4e80fcfe249c800408a5f9d72ebd5933 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 27 Jun 2013 22:00:05 +0200 Subject: ASoC: adau1701: remove control_data assignment codec->control_data has to be left unset to make the ASoC core access the regmap properly. That bug slipped in during a rebase session of the driver refactoring. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 4cd4dd10fa24..d1124a5b3471 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -600,8 +600,6 @@ static int adau1701_probe(struct snd_soc_codec *codec) unsigned int val; struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); - codec->control_data = to_i2c_client(codec->dev); - /* * Let the pll_clkdiv variable default to something that won't happen * at runtime. That way, we can postpone the firmware download from -- cgit v1.2.1 From 05843c07ee501cc89a484b2d02528282838a2318 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Mon, 1 Jul 2013 11:27:16 +0200 Subject: ALSA: hda - Add Dell SSID to support Headset Mic recording This is X5 Precision - Diesel platform. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d6a9f5d2b06..14ac9b0e740c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3585,6 +3585,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3604,6 +3606,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- cgit v1.2.1 From 1ba65ae4bdbd43265c51ee4c30ff21a48124b6d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 3 Jul 2013 14:01:32 +0200 Subject: ALSA: vmaster: Fix the regression of missing vmaster hook call The commit [1ca2f2ec: ALSA: vmaster: Add snd_ctl_sync_vmaster() helper function] changed master_put() function and the check for the required vmaster hook call is wrongly performed now, which results in the missing hook call upon "Master Playback Switch" value changes. This patch corrects the check logic. Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 5df8dc25ad80..842a97d5fc3a 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -348,7 +348,7 @@ static int master_put(struct snd_kcontrol *kcontrol, err = sync_slaves(master, old_val, new_val); if (err < 0) return err; - if (master->hook && first_init) + if (master->hook && !first_init) master->hook(master->hook_private_data, master->val); return 1; } -- cgit v1.2.1