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

[alsa-devel] [PATCH RFC] ALSA: oxygen: Xonar DG(X): fix Stereo Upmixing regression
by Clemens Ladisch 18 Mar '14

18 Mar '14

The code introduced in commit 1f91ecc14dee ("ALSA: oxygen: modify adjust_dg_dac_routing function") accidentally disregarded the old value of the playback routing register, so it broke the "Stereo Upmixing" mixer control. The unmuted parts of the channel routing are the same for all settings of the output destination, so it suffices to revert that part of the patch. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/pci/oxygen/xonar_dg.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) Roman, could you please test this patch, and add your Tested-by? diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index ed6f199..4cf3200 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -238,11 +238,21 @@ void set_cs4245_adc_params(struct oxygen *chip, cs4245_write_spi(chip, CS4245_MCLK_FREQ); } +static inline unsigned int shift_bits(unsigned int value, + unsigned int shift_from, + unsigned int shift_to, + unsigned int mask) +{ + if (shift_from < shift_to) + return (value << (shift_to - shift_from)) & mask; + else + return (value >> (shift_from - shift_to)) & mask; +} + unsigned int adjust_dg_dac_routing(struct oxygen *chip, unsigned int play_routing) { struct dg *data = chip->model_data; - unsigned int routing = 0; switch (data->output_sel) { case PLAYBACK_DST_HP: @@ -252,15 +262,23 @@ unsigned int adjust_dg_dac_routing(struct oxygen *chip, OXYGEN_PLAY_MUTE67, OXYGEN_PLAY_MUTE_MASK); break; case PLAYBACK_DST_MULTICH: - routing = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | - (2 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | - (1 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | - (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, OXYGEN_PLAY_MUTE01, OXYGEN_PLAY_MUTE_MASK); break; } - return routing; + return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC0_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_MASK); } void dump_cs4245_registers(struct oxygen *chip,

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[alsa-devel] Restore default value for alc282
by Kailang 18 Mar '14

18 Mar '14

Hi Takashi, Attachment is the patch for fix below issue for alc282. Restore the registers to prevent the abnormal digital power supply rising ratio/sequence to the codec and causing the incorrect default codec register restoration during initialization. Many Thanks. BR, Kailang

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[alsa-devel] Restore default value for alc283
by Kailang 18 Mar '14

18 Mar '14

Hi Takashi, Attachment is the patch for fix below issue. Restore the registers to prevent the abnormal digital power supply rising ratio/sequence to the codec and causing the incorrect default codec register restoration during initialization. Many Thanks. BR, Kailang

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[alsa-devel] [PATCH] ASoC: tegra: Use flat regcache.
by Dylan Reid 18 Mar '14

18 Mar '14

When using an rbtree cache, there can be allocations the first time a register is accessed. This can cause an attempt to schedule while atomic in the case that the regmap is using a spinlock. This could be fixed by either initializing all the registers or using a flat cache. The register maps for tegra30_ahub and tegra30_i2s are dense and don't save much from using a tree so convert them to flat. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d6f4c99..0db68f4 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -471,7 +471,7 @@ static const struct regmap_config tegra30_ahub_apbif_regmap_config = { .readable_reg = tegra30_ahub_apbif_wr_rd_reg, .volatile_reg = tegra30_ahub_apbif_volatile_reg, .precious_reg = tegra30_ahub_apbif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) @@ -490,7 +490,7 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { .max_register = LAST_REG(AUDIO_RX), .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, .readable_reg = tegra30_ahub_ahub_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static struct tegra30_ahub_soc_data soc_data_tegra30 = { diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 49ad936..f146c41 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -357,7 +357,7 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .writeable_reg = tegra30_i2s_wr_rd_reg, .readable_reg = tegra30_i2s_wr_rd_reg, .volatile_reg = tegra30_i2s_volatile_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static const struct tegra30_i2s_soc_data tegra30_i2s_config = { -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH 1/2] ASoC: sirf-audio-codec: Remove snd_soc_codec_set_cache_io() call
by Lars-Peter Clausen 18 Mar '14

18 Mar '14

There was a overlap between the snd_soc_codec_set_cache_io() cleanup and the addition of the sirf-audio-codec resulting in the sirf-audio-codec driver still using the old signature of snd_soc_codec_set_cache_io(), which will cause a compile error. Since the core is able to automatically setup IO for this driver we can just remove both the snd_soc_set_cache_io() call and the control_data assignment. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sirf-audio-codec.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 90e3a22..58e7c1f 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -337,18 +337,9 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = { static int sirf_audio_codec_probe(struct snd_soc_codec *codec) { - int ret; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); pm_runtime_enable(codec->dev); - codec->control_data = sirf_audio_codec->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; - } if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { snd_soc_dapm_new_controls(dapm, -- 1.8.0

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[alsa-devel] [PATCH v2 0/4] ASoC: simple-card: DT fix and multi DAI links extension
by Jean-Francois Moine 17 Mar '14

17 Mar '14

This patch series fixes a small problem about node device locks and also extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. - v2 - change subject/comment about device node reference count (Mark Brown) - use a null size array instead of an implicit area for the DAI links (Li Xiubo) - update the reference count of the device node at end of probe Jean-Francois Moine (4): ASoC: simple-card: Fix the reference count of device nodes ASoC: simple-card: dynamically allocate the DAI link array ASoC: simple-card: accept many DAI links ASoC: simple-card: Add DT documentation for multi-DAI links .../devicetree/bindings/sound/simple-card.txt | 34 +++- sound/soc/generic/simple-card.c | 174 ++++++++++++++------- 2 files changed, 152 insertions(+), 56 deletions(-) -- 1.9.0

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[alsa-devel] [PATCH] ASoC: fsi: fixup compile error for simple-card
by Kuninori Morimoto 17 Mar '14

17 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patches fixes c7a507eea1db1430476289f525f9c853d5d485e8 (ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags) commit's compie error arch/arm/mach-shmobile/board-mackerel.c:512:2: \ error: unknown field 'fmt' specified in initializer Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark This patch is for mark/fix/rcar arch/arm/mach-shmobile/board-mackerel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 8c43321..0ff4d8e 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -509,7 +509,7 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .fmt = SND_SOC_DAIFMT_CBS_CFS, + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", }, -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Turn power off only once.
by Jyri Sarha 17 Mar '14

17 Mar '14

Regulator code keep count of enables and disables. Double disable causes an ugly warning. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 89f808b..07ddec3 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1021,7 +1021,8 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - aic31xx_power_off(codec); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_power_off(codec); break; } codec->dapm.bias_level = level; -- 1.7.9.5

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[alsa-devel] [PATCH 0/2][RFC] ASoC: fsi/rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags
by Kuninori Morimoto 17 Mar '14

17 Mar '14

Hi Mark, Simon These patches fixup Renesas fsi/rcar driver's SND_SOC_DAIFMT_CBx_CFx flag settings. I had misunderstand these settings... But this is still [RFC], since it modifies DT side code too. >> Mark #1 patch depends on Jyri's "Fix simple-card *-master DT parameter handling" if it was accepted >> Simon Can you check #1 patch which modifys SH-ARM platform ? Kuninori Morimoto (2): 1) ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags 2) ASoC: rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags .../boot/dts/r8a7740-armadillo800eva-reference.dts | 3 ++- arch/arm/boot/dts/sh73a0-kzm9g-reference.dts | 2 ++ arch/arm/mach-shmobile/board-armadillo800eva.c | 4 +--- arch/arm/mach-shmobile/board-kzm9g.c | 4 +--- arch/arm/mach-shmobile/board-mackerel.c | 6 ++---- arch/sh/boards/mach-ecovec24/setup.c | 4 +--- arch/sh/boards/mach-se/7724/setup.c | 4 +--- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 4 ++-- 9 files changed, 13 insertions(+), 20 deletions(-) Best regards --- Kuninori Morimoto

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Re: [alsa-devel] compress: adding support for hardware dependent params
by noman pouigt 16 Mar '14

16 Mar '14

> We have a requirement, where we need to set a number of hardware specific > parameters for compressed offload use-cases, before starting a stream. As > these parameters need to be set once for the entire session, we would > like to explore the best way to set these additional parameters somewhere > before calling trigger() function. > Below are few of the proposed solutions that we could think of at this > point for getting this done- As you have not mentioned what kind of metadata you are talking about I am assuming if your meta data can fit in below structure then we already have a code in upstream which takes care of your need. 140 struct snd_compr_metadata { 141 __u32 key; 142 __u32 value[8]; 143 }; Structure snd_compr_ops has set_metadata and get_metadata callbacks which you can use for your need. 88 * struct snd_compr_ops: compressed path DSP operations 89 * @open: Open the compressed stream 90 * This callback is mandatory and shall keep dsp ready to receive the stream 91 * parameter 92 * @free: Close the compressed stream, mandatory 93 * @set_params: Sets the compressed stream parameters, mandatory 94 * This can be called in during stream creation only to set codec params 95 * and the stream properties 96 * @get_params: retrieve the codec parameters, mandatory 97 * @trigger: Trigger operations like start, pause, resume, drain, stop. 98 * This callback is mandatory 99 * @pointer: Retrieve current h/w pointer information. Mandatory 100 * @copy: Copy the compressed data to/from userspace, Optional 101 * Can't be implemented if DSP supports mmap 102 * @mmap: DSP mmap method to mmap DSP memory 103 * @ack: Ack for DSP when data is written to audio buffer, Optional 104 * Not valid if copy is implemented 105 * @get_caps: Retrieve DSP capabilities, mandatory 106 * @get_codec_caps: Retrieve capabilities for a specific codec, mandatory 107 */ It would be nice if someone can send a patch here to describe set and get_metadata callbacks in compress_driver.h > > > 1. Adding new IOCTL call: Add a new ioctl control in compress-offload.c > file to handle hardware specific parameters setting. This new ioctl > control will call into soc-compress.c and eventually call into platform > driver ioctl to set those hardware specific parameters. In this case, we > will add a new structure, let's say "snd_compr_hw_ctrl_params", with the > same fields as "snd_compr_metadata" struct. > > > 2. Adding a function hw_dependant_hw_params(), which will be called after > hw_params() and before prepare() function. The new function will > eventually call into the platform driver to get the hardware specific > parameters set. Once implemented, this would look something like- > hw_params(); > hw_dependant_hw_params(); > prepare(); > > > 3. Add a function "hw_dependant_hw_params()", as _part_ of the hw_params() > function. This new function will be called after the (existing) generic > parameters are set. So once implemented, the proposed function calls would > look like- > hw_params() { > generic_hw_params(); > hw_dependant_hw_params(); > } > prepare(); > > > We would like to get feedback on these proposed solutions and get a > discussion going to see how best these requirements can be fitted in the > current Compressed ALSA framework (for mainlining). Can you explain your meta data? Be the change you want to see in the world:Gandhi

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