diff options
Diffstat (limited to 'sound')
28 files changed, 386 insertions, 93 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ecdf30eb5879..4aba7646dd9c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg) "Line Out", "Speaker", "HP Out", "CD", "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" + "SPDIF In", "Digital In", "Reserved", "Other" }; return jack_types[(cfg & AC_DEFCFG_DEVICE) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 7dd846380a50..d0d7ac1e99d2 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size) { int i; - int ret; + int ret = 0; int size; /* diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 43c2ea539561..2dbe767be16b 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path); static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - bool changed; + bool changed = false; int i; if (!spec->power_down_unused || path->active) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 418bfc0eb0a3..bcd40ee488e3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " * this may give more power-saving, but will take longer time to * wake up. */ -static int power_save_controller = -1; -module_param(power_save_controller, bint, 0644); +static bool power_save_controller = 1; +module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif /* CONFIG_PM */ @@ -2931,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - if (power_save_controller > 0) - return 0; if (!power_save_controller || !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return -EBUSY; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827d0a95..de8ac5c07fd0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 563c24df4d6f..f15c36bde540 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3440,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) const hda_nid_t *ssids; if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 || + codec->vendor_id == 0x10ec0671) ssids = alc663_ssids; else ssids = alc662_ssids; @@ -3894,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b72561c615..500f666c3875 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC @@ -203,6 +204,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_AK5386 + tristate + config SND_SOC_ALC5623 tristate config SND_SOC_ALC5632 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3b8b41..3a7ec1c39741 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o @@ -137,6 +138,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 068b3ae56a17..1aa10ddf3a61 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -133,6 +133,8 @@ struct adau1373 { #define ADAU1373_DAI_FORMAT_DSP 0x3 #define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_SR_MASK (0x07 << 2) +#define ADAU1373_BCLKDIV_BCLK_MASK 0x03 #define ADAU1373_BCLKDIV_32 0x03 #define ADAU1373_BCLKDIV_64 0x02 #define ADAU1373_BCLKDIV_128 0x01 @@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), - ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, + (div << 2) | ADAU1373_BCLKDIV_64); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 6f6c335a5baa..c7cfdf957e4d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; int ret; @@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, - val); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, + val); if (ret < 0) return ret; @@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - int val = 0; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + int ret, val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ val |= IEC958_AES0_CON_NOT_COPYRIGHT; - snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val); + regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val); val = 0; @@ -132,11 +134,33 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val); + ret = regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val); + if (ret < 0) + return ret; + + /* enable transmitter */ + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); + if (ret < 0) + return ret; + + return 0; +} + +static int ak4104_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + /* disable transmitter */ + return regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, 0); } static const struct snd_soc_dai_ops ak4101_dai_ops = { .hw_params = ak4104_hw_params, + .hw_free = ak4104_hw_free, .set_fmt = ak4104_set_dai_fmt, }; @@ -160,20 +184,17 @@ static int ak4104_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ak4104->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) - return ret; /* set power-up and non-reset bits */ - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); if (ret < 0) return ret; /* enable transmitter */ - ret = snd_soc_update_bits(codec, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); if (ret < 0) return ret; @@ -182,8 +203,10 @@ static int ak4104_probe(struct snd_soc_codec *codec) static int ak4104_remove(struct snd_soc_codec *codec) { - snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); return 0; } diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c new file mode 100644 index 000000000000..1f303983ae02 --- /dev/null +++ b/sound/soc/codecs/ak5386.c @@ -0,0 +1,152 @@ +/* + * ALSA SoC driver for + * Asahi Kasei AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + * + * (c) 2013 Daniel Mack <zonque@gmail.com> + * + * 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 <linux/module.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +struct ak5386_priv { + int reset_gpio; +}; + +static struct snd_soc_codec_driver soc_codec_ak5386; + +static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + format &= SND_SOC_DAIFMT_FORMAT_MASK; + if (format != SND_SOC_DAIFMT_LEFT_J && + format != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int ak5386_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 ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * From the datasheet: + * + * All external clocks (MCLK, SCLK and LRCK) must be present unless + * PDN pin = “L”. If these clocks are not provided, the AK5386 may + * draw excess current due to its use of internal dynamically + * refreshed logic. If the external clocks are not present, place + * the AK5386 in power-down mode (PDN pin = “L”). + */ + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 1); + + return 0; +} + +static int ak5386_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 0); + + return 0; +} + +static const struct snd_soc_dai_ops ak5386_dai_ops = { + .set_fmt = ak5386_set_dai_fmt, + .hw_params = ak5386_hw_params, + .hw_free = ak5386_hw_free, +}; + +static struct snd_soc_dai_driver ak5386_dai = { + .name = "ak5386-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .ops = &ak5386_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id ak5386_dt_ids[] = { + { .compatible = "asahi-kasei,ak5386", }, + { } +}; +MODULE_DEVICE_TABLE(of, ak5386_dt_ids); +#endif + +static int ak5386_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ak5386_priv *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->reset_gpio = -EINVAL; + dev_set_drvdata(dev, priv); + + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) + priv->reset_gpio = of_get_named_gpio(dev->of_node, + "reset-gpio", 0); + + if (gpio_is_valid(priv->reset_gpio)) + if (devm_gpio_request_one(dev, priv->reset_gpio, + GPIOF_OUT_INIT_LOW, + "AK5386 Reset")) + priv->reset_gpio = -EINVAL; + + return snd_soc_register_codec(dev, &soc_codec_ak5386, + &ak5386_dai, 1); +} + +static int ak5386_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak5386_driver = { + .probe = ak5386_probe, + .remove = ak5386_remove, + .driver = { + .name = "ak5386", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ak5386_dt_ids), + }, +}; + +module_platform_driver(ak5386_driver); + +MODULE_DESCRIPTION("ASoC driver for AK5386 ADC"); +MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc176044994d..fc176044994d 100755..100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 7e103f249053..7e103f249053 100755..100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f2d61a187830..566ea3256e2d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: width = SI476X_PCM_FORMAT_S8; + break; case SNDRV_PCM_FORMAT_S16_LE: width = SI476X_PCM_FORMAT_S16_LE; break; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index cb03cc448da6..e895d3939eef 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -581,7 +581,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct arizona *arizona = dev_get_drvdata(codec->dev); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct regmap *regmap = codec->control_data; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 134e41c870b9..f8a31ad0b203 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1083,6 +1083,8 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "ROP", NULL, "Right Speaker PGA" }, { "RON", NULL, "Right Speaker PGA" }, + { "Charge Pump", NULL, "CLK_DSP" }, + { "Left Headphone Output PGA", NULL, "Charge Pump" }, { "Right Headphone Output PGA", NULL, "Charge Pump" }, { "Left Line Output PGA", NULL, "Charge Pump" }, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bc03baef39fa..3470b649c0b2 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -194,17 +194,25 @@ static void wm_adsp_buf_free(struct list_head *list) #define WM_ADSP_NUM_FW 4 +#define WM_ADSP_FW_MBC_VSS 0 +#define WM_ADSP_FW_TX 1 +#define WM_ADSP_FW_TX_SPK 2 +#define WM_ADSP_FW_RX_ANC 3 + static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { - "MBC/VSS", "Tx", "Tx Speaker", "Rx ANC" + [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", + [WM_ADSP_FW_TX] = "Tx", + [WM_ADSP_FW_TX_SPK] = "Tx Speaker", + [WM_ADSP_FW_RX_ANC] = "Rx ANC", }; static struct { const char *file; } wm_adsp_fw[WM_ADSP_NUM_FW] = { - { .file = "mbc-vss" }, - { .file = "tx" }, - { .file = "tx-spk" }, - { .file = "rx-anc" }, + [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, + [WM_ADSP_FW_TX] = { .file = "tx" }, + [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, + [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, @@ -585,13 +593,30 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) buf_size = sizeof(adsp1_id); algs = be32_to_cpu(adsp1_id.algs); + dsp->fw_id = be32_to_cpu(adsp1_id.fw.id); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - be32_to_cpu(adsp1_id.fw.id), + dsp->fw_id, (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp1_id.fw.ver) & 0xff, algs); + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_id.fw.id); + region->base = be32_to_cpu(adsp1_id.zm); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_id.fw.id); + region->base = be32_to_cpu(adsp1_id.dm); + list_add_tail(®ion->list, &dsp->alg_regions); + pos = sizeof(adsp1_id) / 2; term = pos + ((sizeof(*adsp1_alg) * algs) / 2); break; @@ -609,13 +634,38 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) buf_size = sizeof(adsp2_id); algs = be32_to_cpu(adsp2_id.algs); + dsp->fw_id = be32_to_cpu(adsp2_id.fw.id); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - be32_to_cpu(adsp2_id.fw.id), + dsp->fw_id, (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp2_id.fw.ver) & 0xff, algs); + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.xm); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.ym); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.zm); + list_add_tail(®ion->list, &dsp->alg_regions); + pos = sizeof(adsp2_id) / 2; term = pos + ((sizeof(*adsp2_alg) * algs) / 2); break; @@ -817,8 +867,24 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) case (WMFW_INFO_TEXT << 8): break; case (WMFW_ABSOLUTE << 8): - region_name = "register"; - reg = offset; + /* + * Old files may use this for global + * coefficients. + */ + if (le32_to_cpu(blk->id) == dsp->fw_id && + offset == 0) { + region_name = "global coefficients"; + mem = wm_adsp_find_region(dsp, type); + if (!mem) { + adsp_err(dsp, "No ZM\n"); + break; + } + reg = wm_adsp_region_to_reg(mem, 0); + + } else { + region_name = "register"; + reg = offset; + } break; case WMFW_ADSP1_DM: @@ -864,7 +930,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", @@ -901,7 +968,7 @@ out_fw: wm_adsp_buf_free(&buf_list); out: kfree(file); - return 0; + return ret; } int wm_adsp1_init(struct wm_adsp *adsp) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 9f90c9fea842..fea514627526 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -46,6 +46,8 @@ struct wm_adsp { struct list_head alg_regions; + int fw_id; + const struct wm_adsp_region *mem; int num_mems; diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 55464a5b0706..810c7eeb7b03 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_reset) imx_ssi->ac97_reset(ac97); + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) @@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_warm_reset) imx_ssi->ac97_warm_reset(ac97); + + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } struct snd_ac97_bus_ops soc_ac97_ops = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8e52c1485df3..eb4373840bb6 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = { .num_links = ARRAY_SIZE(pcm030_fabric_dai), }; -static int __init pcm030_fabric_probe(struct platform_device *op) +static int pcm030_fabric_probe(struct platform_device *op) { struct device_node *np = op->dev.of_node; struct device_node *platform_np; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e336a7c..6bbeb0bf1a73 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) { struct i2s_dai *i2s; + int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL); if (i2s == NULL) @@ -996,15 +997,17 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.capture.channels_max = 2; i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; + dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ - i2s->pdev = platform_device_register_resndata(NULL, - "samsung-i2s-sec", -1, NULL, 0, NULL, 0); + i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); if (IS_ERR(i2s->pdev)) return NULL; - } - /* Pre-assign snd_soc_dai_set_drvdata */ - dev_set_drvdata(&i2s->pdev->dev, i2s); + platform_set_drvdata(i2s->pdev, i2s); + ret = platform_device_add(i2s->pdev); + if (ret < 0) + return NULL; + } return i2s; } @@ -1107,6 +1110,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (samsung_dai_type == TYPE_SEC) { sec_dai = dev_get_drvdata(&pdev->dev); + if (!sec_dai) { + dev_err(&pdev->dev, "Unable to get drvdata\n"); + return -EFAULT; + } snd_soc_register_dai(&sec_dai->pdev->dev, &sec_dai->i2s_dai_drv); asoc_dma_platform_register(&pdev->dev); diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 19eff8fc4fdd..1a8b03e4b41b 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_platform sh7760_soc_platform = { - .pcm_ops = &camelot_pcm_ops, +static struct snd_soc_platform_driver sh7760_soc_platform = { + .ops = &camelot_pcm_ops, .pcm_new = camelot_pcm_new, .pcm_free = camelot_pcm_free, }; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b5b3db71e253..ed0bfb0ddb96 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -211,19 +211,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { ret = platform->driver->compr_ops->set_params(cstream, params); if (ret < 0) - goto out; + goto err; } if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); if (ret < 0) - goto out; + goto err; } snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); -out: + /* cancel any delayed stream shutdown that is pending */ + rtd->pop_wait = 0; + mutex_unlock(&rtd->pcm_mutex); + + cancel_delayed_work_sync(&rtd->delayed_work); + + return ret; + +err: mutex_unlock(&rtd->pcm_mutex); return ret; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b7e84a7cd9ee..ff4b45a5d796 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2963,7 +2963,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val = val << shift; ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); - if (ret != 0) + if (ret < 0) return ret; if (snd_soc_volsw_is_stereo(mc)) { @@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (params->mask) { ret = regmap_read(codec->control_data, params->base, &val); if (ret != 0) - return ret; + goto out; val &= params->mask; @@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ((u32 *)data)[0] |= cpu_to_be32(val); break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } } ret = regmap_raw_write(codec->control_data, params->base, data, len); +out: kfree(data); return ret; @@ -4197,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4206,7 +4207,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1d6a9b3ceb27..d6d9ba2e6916 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->sink && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_output_ep(path->sink, list); + + path->walking = 0; } } @@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->source && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_input_ep(path->source, list); + + path->walking = 0; } } diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5a1148..5e7aebe1e664 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +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) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + 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 (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + 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) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c925ab0adeb6..5e2c55c5b255 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -43,8 +43,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels_min = 2, @@ -127,26 +125,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_START); - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_STOP); - default: - return -EINVAL; - } - return 0; -} - static int tegra_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { @@ -164,7 +142,7 @@ static struct snd_pcm_ops tegra_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = tegra_pcm_hw_params, .hw_free = tegra_pcm_hw_free, - .trigger = tegra_pcm_trigger, + .trigger = snd_dmaengine_pcm_trigger, .pointer = snd_dmaengine_pcm_pointer, .mmap = tegra_pcm_mmap, }; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2eb282..9e2703a25156 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[4]; - int err, crate; + int err, cur_rate, prev_rate; int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) @@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + prev_rate = 0; + } else { + prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + } + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", dev->devnum, iface, fmt->altsetting); - return err; + cur_rate = 0; + } else { + cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + if (cur_rate != rate) { + snd_printd(KERN_WARNING + "current rate %d is different from the runtime rate %d\n", + cur_rate, rate); + } + + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (rate != prev_rate) { + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + } return 0; } |
