October 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 01/12] ASoC: ak4535: Cleanup manual bias level transitions
by Lars-Peter Clausen 22 Oct '14

22 Oct '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4535.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 30e2978..eced46d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -373,36 +373,19 @@ static struct snd_soc_dai_driver ak4535_dai = { .ops = &ak4535_dai_ops, }; -static int ak4535_suspend(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ak4535_resume(struct snd_soc_codec *codec) { snd_soc_cache_sync(codec); - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } static int ak4535_probe(struct snd_soc_codec *codec) { - /* power on device */ - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); return 0; } -/* power down chip */ -static int ak4535_remove(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static const struct regmap_config ak4535_regmap = { .reg_bits = 8, .val_bits = 8, @@ -417,10 +400,10 @@ static const struct regmap_config ak4535_regmap = { static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .probe = ak4535_probe, - .remove = ak4535_remove, - .suspend = ak4535_suspend, .resume = ak4535_resume, .set_bias_level = ak4535_set_bias_level, + .suspend_bias_off = true, + .dapm_widgets = ak4535_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), .dapm_routes = ak4535_audio_map, -- 1.8.0

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[alsa-devel] [PATCH next 1/2] ALSA: pcm: Remove arch-dependent mmap kludges
by Takashi Iwai 22 Oct '14

22 Oct '14

Since we have consistently dma_mmap_coherent() for all architectures, the current ifdef and arch-specific codes in pcm core can be cleaned up gracefully. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 30 ------------------------------ 1 file changed, 30 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 815396d8427f..aa6754da7929 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -35,9 +35,6 @@ #include <sound/timer.h> #include <sound/minors.h> #include <asm/io.h> -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) -#include <dma-coherence.h> -#endif /* * Compatibility @@ -3251,20 +3248,6 @@ static inline struct page * snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) { void *vaddr = substream->runtime->dma_area + ofs; -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) - return virt_to_page(CAC_ADDR(vaddr)); -#endif -#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { - dma_addr_t addr = substream->runtime->dma_addr + ofs; - addr -= get_dma_offset(substream->dma_buffer.dev.dev); - /* assume dma_handle set via pfn_to_phys() in - * mm/dma-noncoherent.c - */ - return pfn_to_page(addr >> PAGE_SHIFT); - } -#endif return virt_to_page(vaddr); } @@ -3309,13 +3292,6 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .fault = snd_pcm_mmap_data_fault, }; -#ifndef ARCH_HAS_DMA_MMAP_COHERENT -/* This should be defined / handled globally! */ -#if defined(CONFIG_ARM) || defined(CONFIG_ARM64) -#define ARCH_HAS_DMA_MMAP_COHERENT -#endif -#endif - /* * mmap the DMA buffer on RAM */ @@ -3331,7 +3307,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ -#ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3339,11 +3314,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); -#elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV && - !plat_device_is_coherent(substream->dma_buffer.dev.dev)) - area->vm_page_prot = pgprot_noncached(area->vm_page_prot); -#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- 2.1.2

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[alsa-devel] [PATCH 0/2] ASoC: spread usage of snd_ctl_elem_info()
by Takashi Iwai 22 Oct '14

22 Oct '14

Hi, just like my previous patch series, this convers the simple open-code for enum info callbacks with the existing snd_ctl_enum_info() helper. They can be applied individually, as far as I know, but I think it'd fit better to the same branch for other similar patches. Takashi

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[alsa-devel] [PATCH V2 1/2] ARM: pxa: spitz: register spitz-audio device
by Dmitry Eremin-Solenikov 22 Oct '14

22 Oct '14

Register spitz-audio device to be used by ASoC driver. [V2: Removed ifdefs around spitz_audio_init]. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- arch/arm/mach-pxa/spitz.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..af1dea3 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,19 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +static struct platform_device spitz_audio_device = { + .name = "spitz-audio", + .id = -1, +}; + +static inline void spitz_audio_init(void) +{ + platform_device_register(&spitz_audio_device); +} + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +983,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- 2.1.1

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[alsa-devel] [PATCH] ASoC: Intel: Fix block is enabled multiple times issue
by Jie Yang 22 Oct '14

22 Oct '14

During FW parsing and loading, block_list_prepare() may be called for each raw data block copying and this may made the hsw_block_enable() called mutiple times, which increase block->users many times. The result of this is hsw_block_disable() can't power gated the related block when trying to free the blocks during suspend, and the power gating status also confused. Here check the block user status, only calling enable() for those blocks who has no user yet. Remember that this works correctlly on current case, where there are enough SRAM memory so different module won't share a memory block. For further usage, we may need restructure the struct sst_mem_block to save the module list who is using it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index f2f1d5f..db60398 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -80,7 +80,7 @@ static int block_list_prepare(struct sst_dsp *dsp, /* enable each block so that's it'e ready for data */ list_for_each_entry(block, block_list, module_list) { - if (block->ops && block->ops->enable) { + if (block->ops && block->ops->enable && !block->users) { ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: adau1761: Fix input PGA volume
by Lars-Peter Clausen 22 Oct '14

22 Oct '14

For the input PGA to work correctly the ALC clock needs to be active. Otherwise volume changes are not applied. Fixes: dab464b60b2 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/adau1761.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5518ebd..91f6028 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -405,6 +405,7 @@ static const struct snd_soc_dapm_widget adau1761_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Slew Clock", ADAU1761_CLK_ENABLE0, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ALC Clock", ADAU1761_CLK_ENABLE0, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Digital Clock 0", 1, ADAU1761_CLK_ENABLE1, 0, 0, NULL, 0), @@ -436,6 +437,9 @@ static const struct snd_soc_dapm_route adau1761_dapm_routes[] = { { "Right Playback Mixer", NULL, "Slew Clock" }, { "Left Playback Mixer", NULL, "Slew Clock" }, + { "Left Input Mixer", NULL, "ALC Clock" }, + { "Right Input Mixer", NULL, "ALC Clock" }, + { "Digital Clock 0", NULL, "SYSCLK" }, { "Digital Clock 1", NULL, "SYSCLK" }, }; -- 1.8.0

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[alsa-devel] [PATCH 0/5] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 22 Oct '14

22 Oct '14

This patchset implements audio support on Phytec's phyFLEX-i.MX6 boards. This includes creating imx-tlv320aic3x machine driver, fixing tlv320aic3x driver playback speed issue and enabling audio for pbab01 itself.

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[alsa-devel] [PATCH v2 0/2] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 22 Oct '14

22 Oct '14

This patch set adds multi-CODEC support to the simple card. v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (2): ASoC: core: add multi-codec support in DT ASoC: simple-card: add multi-CODECs in DT .../devicetree/bindings/sound/simple-card.txt | 28 +++-- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 114 +++++++++++++-------- sound/soc/soc-core.c | 75 +++++++++++--- 4 files changed, 149 insertions(+), 71 deletions(-) -- 2.1.1

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[alsa-devel] Alsa-lib 32-bit on 64-bit machine
by Alan Horstmann 22 Oct '14

22 Oct '14

Hi, I've tried to find info or examples of this on the web, but to no avail. What is the recommended way to build Alsa-lib on a 64-bit machine to create the additional alternative 32-bit libraries for compatibility with 32-bit applications? The original distro (in this case a Suse Enterprise) has them, but standard './configure make install' only builds the 64-bit variants. I only succeeded by explicitly adding CFLAGS, and wondered if it should work using configure options, or another way? I'm also not sure exactly which flags should be set with that method (m32, march=i686 others?). Does anyone know how it is done in the Suse builds? Thanks, Alan

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[alsa-devel] [RFC PATCH 1/2] ALSA: hda - Fix "PCM" name being used on one DAC when there are two DACs
by David Henningsson 22 Oct '14

22 Oct '14

In the scenario where there is one "Line Out", one "Speaker" and one "Headphone", and there are only two DACs, two outputs will share a DAC. Currently any mixer on such a DAC will get the "PCM" name, which is misleading. Instead use "Headphone+LO" or "Speaker+LO" to better specify what the volume actually controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_generic.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 64220c0..1d742ea 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1038,6 +1038,19 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, break; *index = ch; return "Headphone"; + case AUTO_PIN_LINE_OUT: + /* This deals with the case where we have two DACs and + * one LO, one HP and one Speaker */ + if (!ch && cfg->speaker_outs && cfg->hp_outs) { + bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); + bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + if (hp_lo_shared && spk_lo_shared) + return spec->vmaster_mute.hook ? "PCM" : "Master"; + if (hp_lo_shared) + return "Headphone+LO"; + if (spk_lo_shared) + return "Speaker+LO"; + } } /* for a single channel output, we don't have to name the channel */ -- 1.9.1

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