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author | Linus Torvalds <torvalds@linux-foundation.org> | 2013-07-11 12:45:59 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2013-07-11 12:45:59 -0700 |
commit | 0fb3767b0a5601dd0d528bc8dbefc0567a34b7ec (patch) | |
tree | 76eed01d088080863f5bf97e37ae4ec1adf9d0a5 /sound | |
parent | 0edcd16a4def296bd6492ae0c10a3c4aef9ef7c0 (diff) | |
parent | 42d4ab832d843b5a512b373c86e70caa43a041c8 (diff) | |
download | blackbird-op-linux-0fb3767b0a5601dd0d528bc8dbefc0567a34b7ec.tar.gz blackbird-op-linux-0fb3767b0a5601dd0d528bc8dbefc0567a34b7ec.zip |
Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few small fixes (and cleanups) for HD-audio, USB-audio and ASoC"
* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: fix regression for fixed stream quirk
ALSA: hda - Keep halting ALC5505 DSP
ASoC: wm8962: fix NULL pdata pointer
ASoC: imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found
ASoC: Samsung: Remove redundant comment
ALSA: hda - Fix EAPD vmaster hook for AD1884 & co
ASoC: samsung: Remove obsolete GPIO based DT pinmuxing
ASoC: mxs: register saif mclk to clock framework
Diffstat (limited to 'sound')
-rw-r--r-- | sound/pci/hda/patch_analog.c | 12 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 19 | ||||
-rw-r--r-- | sound/soc/codecs/wm8962.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/imx-sgtl5000.c | 4 | ||||
-rw-r--r-- | sound/soc/mxs/mxs-saif.c | 35 | ||||
-rw-r--r-- | sound/soc/samsung/i2s.c | 66 | ||||
-rw-r--r-- | sound/soc/samsung/s3c-i2s-v2.c | 4 | ||||
-rw-r--r-- | sound/usb/quirks.c | 4 |
8 files changed, 73 insertions, 73 deletions
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 977b0d878dae..d97f0d61a15b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2112,6 +2112,9 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; struct ad198x_spec *spec = codec->spec; + + if (!spec->eapd_nid) + return; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); @@ -3601,13 +3604,16 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, { struct ad198x_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + break; + case HDA_FIXUP_ACT_PROBE: if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) spec->eapd_nid = spec->gen.autocfg.line_out_pins[0]; else spec->eapd_nid = spec->gen.autocfg.speaker_pins[0]; - if (spec->eapd_nid) - spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + break; } } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 14ac9b0e740c..8bd226149868 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -37,6 +37,9 @@ #include "hda_jack.h" #include "hda_generic.h" +/* keep halting ALC5505 DSP, for power saving */ +#define HALT_REALTEK_ALC5505 + /* unsol event tags */ #define ALC_DCVOL_EVENT 0x08 @@ -2659,15 +2662,27 @@ static void alc5505_dsp_init(struct hda_codec *codec) alc5505_coef_set(codec, 0x880c, 0x00000004); /* DRAM Function control */ alc5505_coef_set(codec, 0x880c, 0x00000003); alc5505_coef_set(codec, 0x880c, 0x00000010); + +#ifdef HALT_REALTEK_ALC5505 + alc5505_dsp_halt(codec); +#endif } +#ifdef HALT_REALTEK_ALC5505 +#define alc5505_dsp_suspend(codec) /* NOP */ +#define alc5505_dsp_resume(codec) /* NOP */ +#else +#define alc5505_dsp_suspend(codec) alc5505_dsp_halt(codec) +#define alc5505_dsp_resume(codec) alc5505_dsp_back_from_halt(codec) +#endif + #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); + alc5505_dsp_suspend(codec); return alc_suspend(codec); } @@ -2696,7 +2711,7 @@ static int alc269_resume(struct hda_codec *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); + alc5505_dsp_resume(codec); return 0; } #endif /* CONFIG_PM */ diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b1dc7d426438..e2de9ecfd641 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ 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; int i, trigger, irq_pol; bool dmicclk, dmicdat; diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 7a8bc1220b2e..3f726e4f88db 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -113,13 +113,13 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) 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; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + return -EPROBE_DEFER; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 49d870034bc3..54511c5e6a7c 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <linux/dma-mapping.h> #include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/delay.h> #include <linux/time.h> #include <sound/core.h> @@ -658,6 +659,33 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) return IRQ_HANDLED; } +static int mxs_saif_mclk_init(struct platform_device *pdev) +{ + struct mxs_saif *saif = platform_get_drvdata(pdev); + struct device_node *np = pdev->dev.of_node; + struct clk *clk; + int ret; + + clk = clk_register_divider(&pdev->dev, "mxs_saif_mclk", + __clk_get_name(saif->clk), 0, + saif->base + SAIF_CTRL, + BP_SAIF_CTRL_BITCLK_MULT_RATE, 3, + 0, NULL); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + if (ret == -EEXIST) + return 0; + dev_err(&pdev->dev, "failed to register mclk: %d\n", ret); + return PTR_ERR(clk); + } + + ret = of_clk_add_provider(np, of_clk_src_simple_get, clk); + if (ret) + return ret; + + return 0; +} + static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; @@ -734,6 +762,13 @@ static int mxs_saif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, saif); + /* We only support saif0 being tx and clock master */ + if (saif->id == 0) { + ret = mxs_saif_mclk_init(pdev); + if (ret) + dev_warn(&pdev->dev, "failed to init clocks\n"); + } + ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, &mxs_saif_dai, 1); if (ret) { diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 82ebb1a51479..7a1734697434 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1016,52 +1016,6 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } -#ifdef CONFIG_OF -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) -{ - struct device *dev = &i2s->pdev->dev; - int index, gpio, ret; - - for (index = 0; index < 7; index++) { - gpio = of_get_gpio(dev->of_node, index); - if (!gpio_is_valid(gpio)) { - dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); - goto free_gpio; - } - - ret = gpio_request(gpio, dev_name(dev)); - if (ret) { - dev_err(dev, "gpio [%d] request failed\n", gpio); - goto free_gpio; - } - i2s->gpios[index] = gpio; - } - return 0; - -free_gpio: - while (--index >= 0) - gpio_free(i2s->gpios[index]); - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) -{ - unsigned int index; - for (index = 0; index < 7; index++) - gpio_free(i2s->gpios[index]); -} -#else -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) -{ - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) -{ -} - -#endif - static const struct of_device_id exynos_i2s_match[]; static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) @@ -1235,18 +1189,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->sec_dai = sec_dai; } - if (np) { - if (samsung_i2s_parse_dt_gpio(pri_dai)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } - } else { - if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } + if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; } snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component, @@ -1267,14 +1213,10 @@ static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; struct resource *res; - struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; - if (!i2s_pdata->cfg_gpio && pdev->dev.of_node) - samsung_i2s_dt_gpio_free(i2s->pri_dai); - if (other) { other->pri_dai = NULL; other->sec_dai = NULL; diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 20e98d1dded2..e5e81b111001 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -1,6 +1,4 @@ -/* sound/soc/samsung/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. +/* ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. * * Copyright (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5b01330b8452..1bc45e71f1fe 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -129,6 +129,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, { struct audioformat *fp; struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; int stream, err; unsigned *rate_table = NULL; @@ -166,6 +167,9 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + altsd = get_iface_desc(alts); + fp->protocol = altsd->bInterfaceProtocol; + if (fp->datainterval == 0) fp->datainterval = snd_usb_parse_datainterval(chip, alts); if (fp->maxpacksize == 0) |