December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate
by Takashi Iwai 18 Dec '13

18 Dec '13

So far, CONFIG_SND_HDA_CODEC_* kconfigs have been booleans due to historical reasons. The major reason was that the automatic codec driver probing wouldn't work if user sets a codec driver as a module while the controller driver as a built-in. And, another reason was to avoid exporting symbols of the helper codes when all drivers are built in. But, this sort of "kindness" rather confuses people in the end, especially makes the config refinement via localmodconfig unhappy. Also, a codec module would still work if you re-bind the controller driver via sysfs (although it's no automatic loading), so there might be a slight use case. That said, better to let people fallen into a pitfall than being too smart and restrict something. Let's make things straightforward: now all CONFIG_SND_HDA_CODEC_* become tristate, and all symbols exported unconditionally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 115 ++++++++++------------------------------------ sound/pci/hda/Makefile | 50 ++++++-------------- sound/pci/hda/hda_codec.h | 13 ------ 3 files changed, 37 insertions(+), 141 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 4cdd9ded4563..b0c0481c3339 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -87,126 +87,72 @@ config SND_HDA_PATCH_LOADER This option turns on hwdep and reconfig features automatically. config SND_HDA_CODEC_REALTEK - bool "Build Realtek HD-audio codec support" - default y + tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include Realtek HD-audio codec support in + Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-realtek. - This module is automatically loaded at probing. - config SND_HDA_CODEC_ANALOG - bool "Build Analog Device HD-audio codec support" - default y + tristate "Build Analog Device HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include Analog Device HD-audio codec support in + Say Y or M here to include Analog Device HD-audio codec support in snd-hda-intel driver, such as AD1986A. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-analog. - This module is automatically loaded at probing. - config SND_HDA_CODEC_SIGMATEL - bool "Build IDT/Sigmatel HD-audio codec support" - default y + tristate "Build IDT/Sigmatel HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include IDT (Sigmatel) HD-audio codec support in + Say Y or M here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-idt. - This module is automatically loaded at probing. - config SND_HDA_CODEC_VIA - bool "Build VIA HD-audio codec support" - default y + tristate "Build VIA HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include VIA HD-audio codec support in + Say Y or M here to include VIA HD-audio codec support in snd-hda-intel driver, such as VT1708. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-via. - This module is automatically loaded at probing. - config SND_HDA_CODEC_HDMI - bool "Build HDMI/DisplayPort HD-audio codec support" - default y + tristate "Build HDMI/DisplayPort HD-audio codec support" help - Say Y here to include HDMI and DisplayPort HD-audio codec + Say Y or M here to include HDMI and DisplayPort HD-audio codec support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-hdmi. - This module is automatically loaded at probing. - config SND_HDA_I915 bool default y depends on DRM_I915 config SND_HDA_CODEC_CIRRUS - bool "Build Cirrus Logic codec support" - default y + tristate "Build Cirrus Logic codec support" select SND_HDA_GENERIC help - Say Y here to include Cirrus Logic codec support in + Say Y or M here to include Cirrus Logic codec support in snd-hda-intel driver, such as CS4206. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-cirrus. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CONEXANT - bool "Build Conexant HD-audio codec support" - default y + tristate "Build Conexant HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include Conexant HD-audio codec support in + Say Y or M here to include Conexant HD-audio codec support in snd-hda-intel driver, such as CX20549. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-conexant. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CA0110 - bool "Build Creative CA0110-IBG codec support" - default y + tristate "Build Creative CA0110-IBG codec support" select SND_HDA_GENERIC help - Say Y here to include Creative CA0110-IBG codec support in + Say Y or M here to include Creative CA0110-IBG codec support in snd-hda-intel driver, found on some Creative X-Fi cards. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-ca0110. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CA0132 - bool "Build Creative CA0132 codec support" - default y + tristate "Build Creative CA0132 codec support" help - Say Y here to include Creative CA0132 codec support in + Say Y or M here to include Creative CA0132 codec support in snd-hda-intel driver. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-ca0132. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 @@ -220,35 +166,22 @@ config SND_HDA_CODEC_CA0132_DSP (ctefx.bin). config SND_HDA_CODEC_CMEDIA - bool "Build C-Media HD-audio codec support" - default y + tristate "Build C-Media HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include C-Media HD-audio codec support in + Say Y or M here to include C-Media HD-audio codec support in snd-hda-intel driver, such as CMI9880. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-cmedia. - This module is automatically loaded at probing. - config SND_HDA_CODEC_SI3054 - bool "Build Silicon Labs 3054 HD-modem codec support" - default y + tristate "Build Silicon Labs 3054 HD-modem codec support" help - Say Y here to include Silicon Labs 3054 HD-modem codec + Say Y or M here to include Silicon Labs 3054 HD-modem codec (and compatibles) support in snd-hda-intel driver. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-si3054. - This module is automatically loaded at probing. - config SND_HDA_GENERIC - bool "Enable generic HD-audio codec parser" - default y + tristate "Enable generic HD-audio codec parser" help - Say Y here to enable the generic HD-audio codec parser + Say Y or M here to enable the generic HD-audio codec parser in snd-hda-intel driver. config SND_HDA_POWER_SAVE_DEFAULT diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 5a40c652df41..1fcb118e480a 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -27,43 +27,19 @@ snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o # common driver obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o -# codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans) -ifdef CONFIG_SND_HDA_GENERIC -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-generic.o -endif -ifdef CONFIG_SND_HDA_CODEC_REALTEK -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o -endif -ifdef CONFIG_SND_HDA_CODEC_CMEDIA -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cmedia.o -endif -ifdef CONFIG_SND_HDA_CODEC_ANALOG -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-analog.o -endif -ifdef CONFIG_SND_HDA_CODEC_SIGMATEL -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-idt.o -endif -ifdef CONFIG_SND_HDA_CODEC_SI3054 -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o -endif -ifdef CONFIG_SND_HDA_CODEC_CIRRUS -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o -endif -ifdef CONFIG_SND_HDA_CODEC_CA0110 -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o -endif -ifdef CONFIG_SND_HDA_CODEC_CA0132 -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o -endif -ifdef CONFIG_SND_HDA_CODEC_CONEXANT -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o -endif -ifdef CONFIG_SND_HDA_CODEC_VIA -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o -endif -ifdef CONFIG_SND_HDA_CODEC_HDMI -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-hdmi.o -endif +# codec drivers +obj-$(CONFIG_SND_HDA_GENERIC) += snd-hda-codec-generic.o +obj-$(CONFIG_SND_HDA_CODEC_REALTEK) += snd-hda-codec-realtek.o +obj-$(CONFIG_SND_HDA_CODEC_CMEDIA) += snd-hda-codec-cmedia.o +obj-$(CONFIG_SND_HDA_CODEC_ANALOG) += snd-hda-codec-analog.o +obj-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += snd-hda-codec-idt.o +obj-$(CONFIG_SND_HDA_CODEC_SI3054) += snd-hda-codec-si3054.o +obj-$(CONFIG_SND_HDA_CODEC_CIRRUS) += snd-hda-codec-cirrus.o +obj-$(CONFIG_SND_HDA_CODEC_CA0110) += snd-hda-codec-ca0110.o +obj-$(CONFIG_SND_HDA_CODEC_CA0132) += snd-hda-codec-ca0132.o +obj-$(CONFIG_SND_HDA_CODEC_CONEXANT) += snd-hda-codec-conexant.o +obj-$(CONFIG_SND_HDA_CODEC_VIA) += snd-hda-codec-via.o +obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 3c484c27048b..2b5d19e48a27 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -681,19 +681,6 @@ snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec, struct snd_dma_buffer *dmab) {} #endif -/* - * Codec modularization - */ - -/* Export symbols only for communication with codec drivers; - * When built in kernel, all HD-audio drivers are supposed to be statically - * linked to the kernel. Thus, the symbols don't have to (or shouldn't) be - * exported unless it's built as a module. - */ -#ifdef MODULE #define EXPORT_SYMBOL_HDA(sym) EXPORT_SYMBOL_GPL(sym) -#else -#define EXPORT_SYMBOL_HDA(sym) -#endif #endif /* __SOUND_HDA_CODEC_H */ -- 1.8.5

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[alsa-devel] [RFC][PATCH 00/17] Enhancement for firewire-lib
by Takashi Sakamoto 18 Dec '13

18 Dec '13

This series of patch extend functionalities of current firewire-lib: - support for both in/out AMDTP stream - support for duplex stream synchronization (blocking mode only) - support for both MIDI/PCM stream - support for both in/out CMP - support for some AV/C general commands Additionally, at the last of this series, some quirks for Fireworks are added. I'm happy to receive your comments, especially for sort processing. Takashi Sakamoto (17): firewire-lib: Rename functions, structure, member for both direction firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse in future firewire-lib: Add support for AMDTP transmit stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: give syt value to handle_out_packet() firewire-lib: Add support for duplex streams with synchronization firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks chipset sound/firewire/amdtp.c | 1014 +++++++++++++++++++++++++++++++++++---------- sound/firewire/amdtp.h | 161 +++++-- sound/firewire/cmp.c | 217 +++++++--- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 149 +++++++ sound/firewire/fcp.h | 16 + sound/firewire/speakers.c | 94 ++--- 8 files changed, 1319 insertions(+), 392 deletions(-) -- 1.8.3.2

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[alsa-devel] [RFC PATCH] ASoC: wm_adsp: Change ADSPs to be mixer widgets rather the PGAs
by Charles Keepax 18 Dec '13

18 Dec '13

PGAs often control the output from a chip and if the DSP is also marked as a PGA it may be powered up after the output has been enabled. This patch changes the ADSP2 cores to be marked as snd_soc_dapm_mixer widgets so they are powered up before any PGAs. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi, Was also considering if it would be worth adding an additional snd_soc_dapm_dsp id? That could sit between mixers and pgas, but I can't really see any obvious issue with treating the DSP as a mixer and it is a much simpler change. Although I am open to writing the other change if it is preferred? Thanks, Charles sound/soc/codecs/wm_adsp.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index d018dea..795a3b2 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -66,7 +66,7 @@ struct wm_adsp { wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) #define WM_ADSP2(wname, num) \ - SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ + SND_SOC_DAPM_MIXER_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[]; -- 1.7.2.5

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[alsa-devel] [PATCH v3] ASoC: fsl: imx-wm8962: Grant hw_params/free() permission to control FLL
by Nicolin Chen 18 Dec '13

18 Dec '13

From: Nicolin Chen <b42378(a)freescale.com> Previously, we couldn't use hw_params() and hw_free() to open and close FLL becuase there might be a race between two simmultaneous substreams so the FLL configuration would be changed and accordingly mulfunction. Also it wouldn't make sense for bypass path feature of WM8962. So we adopted DAPM way to control it. However, if we want to playback a different sample rate file, we need to wait for DAPM to change its bias_level and reconfigure FLL. But after we introduced full symmetry protection in the soc-pcm, we don't need to worry about the race any more. And the instance by using hw_params() and hw_free() to control FLL will allow us to support flexible use cases, 'aplay -Dhw:0 44k16bit.wav 48k24bit.wav 32k16bit.wav' for example. Thus this patch mainly adds FLL configuration code to hw_params/hw_free() so as to enchance the sound card's capability. Meanwhile in order not to break the bypass path feature, we make both set_bias_level() and hw_xxx() ways coexist. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v3: * Added analogue bypass path check in hw_free() to protect bypass route. * Kept sample_rate and sample_format to global static. v2: * Added spinlock potection for boolean fll_locked in PATCH-2 sound/soc/fsl/imx-wm8962.c | 154 +++++++++++++++++++++++++++++++-------------- 1 file changed, 108 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc..d6aa854 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -17,6 +17,7 @@ #include <linux/of_platform.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -35,6 +36,8 @@ struct imx_wm8962_data { char platform_name[DAI_NAME_SIZE]; struct clk *codec_clk; unsigned int clk_frequency; + spinlock_t fll_lock; + bool fll_locked; }; struct imx_priv { @@ -51,18 +54,114 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { static int sample_rate = 44100; static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; +static int imx_wm8962_enable_fll(struct snd_soc_dai *codec_dai, u32 rate, + snd_pcm_format_t format) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned long flags; + u32 freq, ret; + + spin_lock_irqsave(&data->fll_lock, flags); + if (data->fll_locked) { + spin_unlock_irqrestore(&data->fll_lock, flags); + return 0; + } + + data->fll_locked = true; + spin_unlock_irqrestore(&data->fll_lock, flags); + + if (format == SNDRV_PCM_FORMAT_S24_LE) + freq = rate * 384; + else + freq = rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, freq); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + freq, SND_SOC_CLOCK_IN); + if (ret) + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + + return ret; +} + +static int imx_wm8962_disable_fll(struct snd_soc_dai *codec_dai) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned long flags; + int ret; + + spin_lock_irqsave(&data->fll_lock, flags); + if (!data->fll_locked) { + spin_unlock_irqrestore(&data->fll_lock, flags); + return 0; + } + + data->fll_locked = false; + spin_unlock_irqrestore(&data->fll_lock, flags); + + /* Switch to MCLK as sysclk once so as to disable FLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 0, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + /* Disable FLL so that we can reset its output freq later */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 0, 0); + if (ret) + dev_err(dev, "failed to stop FLL: %d\n", ret); + + return ret; +} static int imx_hifi_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; + sample_rate = params_rate(params); sample_format = params_format(params); - return 0; + return imx_wm8962_enable_fll(codec_dai, sample_rate, sample_format); +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + u32 mask, bypass = 0; + + mask = WM8962_MIXINL_TO_HPMIXL_MASK | WM8962_MIXINR_TO_HPMIXL_MASK | + WM8962_IN4L_TO_HPMIXL_MASK | WM8962_IN4R_TO_HPMIXL_MASK; + bypass |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_1) & mask; + bypass |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_2) & mask; + bypass |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_1) & mask; + bypass |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_2) & mask; + + /* Don't diable FLL if running multi-substreams or analogue bypass */ + if (codec_dai->active != 1 || bypass) + return 0; + + return imx_wm8962_disable_fll(codec_dai); } static struct snd_soc_ops imx_hifi_ops = { .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, }; static int imx_wm8962_set_bias_level(struct snd_soc_card *card, @@ -70,60 +169,20 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, 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; - } - } + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + return imx_wm8962_enable_fll(codec_dai, sample_rate, + sample_format); 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; - } - } + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + return imx_wm8962_disable_fll(codec_dai); break; default: @@ -225,6 +284,9 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } + data->fll_locked = false; + spin_lock_init(&data->fll_lock); + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); -- 1.8.4

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[alsa-devel] [PATCH v2 RESEND 1/2] ASOC: tegra: fix matching of AC97 components
by Lucas Stach 18 Dec '13

18 Dec '13

Matching works completely based on the cpu of_node. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> Acked-by: Stephen Warren <swarren(a)nvidia.com> --- v2: no changes CC: <alsa-devel(a)alsa-project.org> CC: Mark Brown <broonie(a)linaro.org> CC: Stephen Warren <swarren(a)wwwdotorg.org> --- sound/soc/tegra/tegra_wm9712.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 5e11963..45b5789 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: wm8904: fix DSP mode B configuration
by Bo Shen 18 Dec '13

18 Dec '13

When wm8904 work in DSP mode B, we still need to configure it to work in DSP mode. Or else, it will work in Right Justified mode. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3938fb1..53bbfac 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: - aif1 |= WM8904_AIF_LRCLK_INV; + aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Remove duplicate info message for DSP RAM ready
by Charles Keepax 18 Dec '13

18 Dec '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index de47ad7..456784f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1489,7 +1489,6 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) } adsp_dbg(dsp, "RAM ready after %d polls\n", count); - adsp_info(dsp, "RAM ready after %d polls\n", count); return 0; } -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: mmp-pcm: config pcm slave via generic dmaengine
by Qiao Zhou 18 Dec '13

18 Dec '13

use snd_dmaengine_pcm_prepare_slave_config to set slave config, and remove the max_burst_size = 4 hard code. select SND_SOC_GENERIC_DMAENGINE_PCM for mmp-pcm. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 18 +++--------------- 2 files changed, 4 insertions(+), 16 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4db74a0..6473052 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,7 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP - select SND_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM help Say Y if you want to add support for codecs attached to diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 7929e19..682ee52 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -67,27 +67,15 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma_params; struct dma_slave_config slave_config; int ret; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!dma_params) - return 0; - - ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + ret = + snd_dmaengine_pcm_prepare_slave_config(substream, params, + &slave_config); if (ret) return ret; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->addr; - slave_config.dst_maxburst = 4; - } else { - slave_config.src_addr = dma_params->addr; - slave_config.src_maxburst = 4; - } - ret = dmaengine_slave_config(chan, &slave_config); if (ret) return ret; -- 1.7.0.4

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[alsa-devel] [RFC PATCH] ALSA: hda - Explicitly keep codec powered up in hdmi_present_sense
by David Henningsson 18 Dec '13

18 Dec '13

This should help us avoid the following mutex deadlock: [] mutex_lock+0x2a/0x50 [] hdmi_present_sense+0x53/0x3a0 [snd_hda_codec_hdmi] [] generic_hdmi_resume+0x5a/0x70 [snd_hda_codec_hdmi] [] hda_call_codec_resume+0xec/0x1d0 [snd_hda_codec] [] snd_hda_power_save+0x1e4/0x280 [snd_hda_codec] [] codec_exec_verb+0x5f/0x290 [snd_hda_codec] [] snd_hda_codec_read+0x5b/0x90 [snd_hda_codec] [] snd_hdmi_get_eld_size+0x1e/0x20 [snd_hda_codec_hdmi] [] snd_hdmi_get_eld+0x2c/0xd0 [snd_hda_codec_hdmi] [] hdmi_present_sense+0x9a/0x3a0 [snd_hda_codec_hdmi] [] hdmi_repoll_eld+0x34/0x50 [snd_hda_codec_hdmi] Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) So; I came across the above stack trace when testing the latest daily HDA driver. It only happened once, and jack polling was enabled while testing, if that matters. As you can see, the hdmi_present_sense is called recursively, and my patch doesn't really fix that; it should only make the consequences less severe (because now the recursion would be in the very beginning, i e, at snd_hda_power_up). I first thought just unlocking the mutex before calling pin_get_eld could do, but not sure if we really want to do that. Nevertheless, I haven't done much of the power management work anyway, so I'd like you to double check it and potentially come up with better ideas if you have any. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f5060fc..977db17 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1496,11 +1496,14 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * specification worked this way. Hence, we just ignore the data in * the unsolicited response to avoid custom WARs. */ - int present = snd_hda_pin_sense(codec, pin_nid); + int present; bool update_eld = false; bool eld_changed = false; bool ret; + snd_hda_power_up(codec); + present = snd_hda_pin_sense(codec, pin_nid); + mutex_lock(&per_pin->lock); pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); if (pin_eld->monitor_present) @@ -1573,6 +1576,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) jack->block_report = !ret; mutex_unlock(&per_pin->lock); + snd_hda_power_down(codec); return ret; } -- 1.7.9.5

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[alsa-devel] [V2 PATCH] ALSA: hda - Add Dell headset detection quirk for one more laptop model
by Hui Wang 18 Dec '13

18 Dec '13

On the Dell machines with codec whose Subsystem Id is 0x10280640, no external microphone can be detected when plugging a 3-ring headset. Using ALC255_FIXUP_DELL1_MIC_NO_PRESENCE can fix this problem. The codec (Vendor ID: 0x10ec0255) on the machine belongs to alc_269 family. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3578f11..5b96698 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4309,6 +4309,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_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), -- 1.8.1.2

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