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

[alsa-devel] [PATCH v3 1/2] mfd: arizona: Add support for INn_Modec register control
by Inha Song 29 Aug '14

29 Aug '14

Some boards need to set the INn_MODE[1:0] register to change the input signal patch. This wlf,inmode property is optional. If present, values must be specified less than or equal to the number of input singals. If values less than the number of input signals, elements that has not been specifed are set to 0 by default. Example: - wlf,inmode = <2 0 2>; /* IN1, IN3 use DMIC */ Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/arizona-core.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 10a0cb9..81e6db8 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -534,7 +534,7 @@ EXPORT_SYMBOL_GPL(arizona_of_get_named_gpio); static int arizona_of_get_core_pdata(struct arizona *arizona) { struct arizona_pdata *pdata = &arizona->pdata; - int ret, i; + int ret, i, proplen; pdata->reset = arizona_of_get_named_gpio(arizona, "wlf,reset", true); @@ -560,6 +560,12 @@ static int arizona_of_get_core_pdata(struct arizona *arizona) ret); } + proplen = of_property_count_u32_elems(arizona->dev->of_node, + "wlf,inmode"); + if (proplen > 0 && proplen <= ARRAY_SIZE(arizona->pdata.inmode)) + of_property_read_u32_array(arizona->dev->of_node, "wlf,inmode", + arizona->pdata.inmode, proplen); + return 0; } -- 2.0.0.390.gcb682f8

3 5

[alsa-devel] [PATCH 0/2] Fix a regression for Dice driver at higher sampling rate
by Takashi Sakamoto 29 Aug '14

29 Aug '14

Commit 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") brought a regression to Dice driver. As a result, Dice driver cannot work correctly at higher sampling rate. This series of patches fixes the regression. Reported-by: Daniel Robbins <drobbins(a)funtoo.org> Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") Takashi Sakamoto (2): ALSA: dice: fix wrong channel mappping at higher sampling rate ALSA: firewire-lib/dice: add arrangements of PCM pointer and interrupts for Dice quirk sound/firewire/amdtp.c | 11 ++++++++++- sound/firewire/amdtp.h | 1 + sound/firewire/dice.c | 29 ++++++++++++++++++++--------- 3 files changed, 31 insertions(+), 10 deletions(-) -- 1.9.1

2 3

Re: [alsa-devel] [PATCH 8/8] ASoC: add snd-soc-dummy DT support
by Kuninori Morimoto 28 Aug '14

28 Aug '14

Hi Lars Thank you for your comment > > sound { > > compatible = "simple-audio-card"; > > > > /* FrontEnd */ > > simple-audio-card,dai-link@0 { > > ... > > format = "left_j"; > > remote = <&endpoint>; > > > > cpu { > > sound-dai = <&rcar_sound 0>; > > }; > > codec { /* dummy */ }; > > }; > > > > /* BackEnd */ > > endpoint: simple-audio-card,dai-link@1 { > > ... > > format = "left_j"; > > > > cpu { /* dummy */ }; > > codec1: codec { > > sound-dai = <&ak4643>; > > }; > > }; > > }; > > When you try to come up with with a binding try to completely ignore that > something call DPCM exists. The binding is supposed to describe the hardware > and how the different hardware components are interconnected. So try to come > up with a binding that accurately describes the hardware connections. Once > that is done try to map the binding onto the existing software framework. > The last step may require some adjustments to the framework. Now, my system is working well with simple-card by this sound { compatible = "simple-audio-card"; ... cpu { sound-dai = <&rcar_sound 0>; }; codec { sound-dai = <&ak4643>; }; }; The reason why I'm tring to support DPCM on simple-card is "sampling rate convert". My rcar_sound can convert sampling rate, and I tried to add this feature as rcar_sound property. But, Mark rejected and requests me to use DPCM for it, since it can be common featrue. Current existing simple-card can't use it, and I'm tring. But, am I misunderstanding ? Best regards --- Kuninori Morimoto

3 5

[alsa-devel] [PATCH] ASoC: cs35l32: Simplify implementation of cs35l32_codec_set_sysclk
by Axel Lin 28 Aug '14

28 Aug '14

Use single snd_soc_update_bits() call to update the register bits. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.c | 28 +++++++--------------------- 1 file changed, 7 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 9c6b272..380c212 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -242,41 +242,27 @@ static struct snd_soc_dai_driver cs35l32_dai[] = { static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { + unsigned int val; switch (freq) { case 6000000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, 0); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, - CS35L32_MCLK_RATIO); + val = CS35L32_MCLK_RATIO; break; case 12000000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, - CS35L32_MCLK_DIV2_MASK); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, - CS35L32_MCLK_RATIO); + val = CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO; break; case 6144000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, 0); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, 0); + val = 0; break; case 12288000: - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_DIV2_MASK, - CS35L32_MCLK_DIV2_MASK); - snd_soc_update_bits(codec, CS35L32_CLK_CTL, - CS35L32_MCLK_RATIO_MASK, 0); + val = CS35L32_MCLK_DIV2_MASK; break; default: return -EINVAL; } - return 0; + return snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO_MASK, val); } static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { -- 1.9.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: cs35l32: Remove unneeded regulator_bulk_free call in cs35l32_i2c_remove
by Axel Lin 28 Aug '14

28 Aug '14

The regulator_bulk_free() call is not required because current code is using devm_regulator_bulk_get(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 380c212..718569f 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -548,8 +548,6 @@ static int cs35l32_i2c_remove(struct i2c_client *i2c_client) if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); - regulator_bulk_free(ARRAY_SIZE(cs35l32->supplies), cs35l32->supplies); - return 0; } -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: Remove unused cache_only from struct snd_soc_codec
by Jarkko Nikula 28 Aug '14

28 Aug '14

There are no real users for cache_only in "struct snd_soc_codec" so remove it and needless debugfs node. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 2 -- 2 files changed, 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e43fbb6f8f8c..29650de0df7b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -792,7 +792,6 @@ struct snd_soc_codec { unsigned int ac97_registered:1; /* Codec has been AC97 registered */ unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int cache_init:1; /* codec cache has been initialized */ - u32 cache_only; /* Suppress writes to hardware */ u32 cache_sync; /* Cache needs to be synced to hardware */ /* codec IO */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4974b4f9c952..b1327ef05860 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -311,8 +311,6 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->component.debugfs_root, - &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->component.debugfs_root, -- 2.1.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: cs35l32: use true/false returns for bool functions
by Brian Austin 28 Aug '14

28 Aug '14

Return true or false instead of 1 and 0 Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs35l32.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 9c6b272..ca897c4 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -121,9 +121,9 @@ static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) case CS35L32_INT_STATUS_2: case CS35L32_INT_STATUS_3: case CS35L32_LED_STATUS: - return 1; + return true; default: - return 0; + return false; } } @@ -134,9 +134,9 @@ static bool cs35l32_precious_register(struct device *dev, unsigned int reg) case CS35L32_INT_STATUS_2: case CS35L32_INT_STATUS_3: case CS35L32_LED_STATUS: - return 1; + return true; default: - return 0; + return false; } } -- 1.7.9.5

2 3

[alsa-devel] [PATCH 1/3] Add CHIP_ID as a readable register.
by Paul Handrigan 28 Aug '14

28 Aug '14

Reported-by: Zoltán Szenczi <zoltan(a)raspberrypi.org> Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs4265.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30c..2dad15a 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -77,6 +77,7 @@ static bool cs4265_readable_register(struct device *dev, unsigned int reg) case CS4265_INT_MASK: case CS4265_STATUS_MODE_MSB: case CS4265_STATUS_MODE_LSB: + case CS4265_CHIP_ID: return true; default: return false; -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: Allow SND_SOC_WM8978 to be selected manually
by Geert Uytterhoeven 28 Aug '14

28 Aug '14

When using a DT-based multi-platform kernel, there's not always Kconfig logic that selects the right codec driver. Allow the user to manually select WM8978. This is needed for Armadillo 800 EVA using a generic r8a7740 multi-platform kernel. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7678122f8fe0..8ab154756934 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -731,7 +731,8 @@ config SND_SOC_WM8974 tristate config SND_SOC_WM8978 - tristate + tristate "Wolfson Microelectronics WM8978 codec" + depends on I2C config SND_SOC_WM8983 tristate -- 1.9.1

2 1

Re: [alsa-devel] [question about DPCM] dpcm_be_disconnect race condition
by Mark Brown 28 Aug '14

28 Aug '14

On Wed, Aug 27, 2014 at 04:15:00PM +0800, Qiao Zhou wrote: > Hi Liam, Mark > > I'm developing audio with DPCM on one phone platform, and met one kernel > panic issue related with dpcm_be_connect. Could you please help to suggest? Adding Liam. > Taking audio playback for example: > 1. User space writes all the data into audio buffer. When the blocking write > returns, it closes audio path(which updates dapm and triggers > soc_dpcm_runtime_update). > 2. When user space closes audio path, kernel audio driver(DMA) is still > transferring data from audio buffer to audio peripheral. So DMA interrupt > will come, and the interrupt handler runs on different cpu comparing to > which the task to close audio path runs on. > 3. Due to system schedule, below case may happen. > > cpu0 cpu1 > > soc_dapm_mixer_update_power -> | DMA interrupt -> > soc_dpcm_runtime_update -> | snd_pcm_period_elapsed -> > dpcm_be_disconnect(old) -> | snd_pcm_update_hw_ptr0 -> > __________________________ | xrun -> > | | | snd_pcm_stop -> > |list_del(&dpcm->list_be) | | dpcm_fe_dai_trigger -> > |list_del(&dpcm->list_fe) | | dpcm_be_dai_trigger -> > |kfree(dpcm); | | list_for_each_entry(dpcm.. > |_________________________| | > > On CPU1, list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) > will get dpcm and corresponding be, but there is no protection against the > dpcm_be_disconnect on cpu0. Related dpcm for the BE may be invalid/freed > when cpu1 tries to access it. > Should we require user space to make sure audio transfer ends before closing > audio path to avoid such issue? I'm not sure if I missed anything or used it > wrongly. Well, it should do that in general if it wants to make sure that the audio actually gets played but the kernel should protect against this. We should be holding a lock whenever we update the lists. > Thanks in advance! > -- > > Best Regards > Qiao >

2 2