December 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] speaker-test: Fix chmap wav file selection.
by Arnaud Pouliquen 07 Dec '16

07 Dec '16

The channel selection currently does not work properly when a channel map control is provided but no manual channel map was explicitly requested with "-m". For example, the CEA/HDMI 6ch (surround 5.1) map is: FL, FR, LFE, FC, RL, RR. Tested command: speaker-test -D hdmi -c 6 -t wav Speaker-test tries to play channels in this following order: 0 - Front Left 3 - Front Center 1 - Front Right 5 - Rear Right 4 - Rear Left 2 - LFE But wav file played on associated speakers are not aligned. Here are the real files played: 0- /usr/share/sounds/alsa/Front_Left.wav => OK 3- /usr/share/sounds/alsa/Rear_Right.wav => OK 1- /usr/share/sounds/alsa/Front_Right.wav => OK 5- /usr/share/sounds/alsa/Rear_Center.wav => KO 4- found file /usr/share/sounds/alsa/Front_Center.wav => KO 2- /usr/share/sounds/alsa/Rear_Left.wav => KO Issue is that associated wav files ordering is reworked only if channel_map_set variable is set. Fix consists in allowing wavs re-ordering if a channel mapping as been get or set, i.e. channel_map is not null. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- speaker-test/speaker-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 5b9cbec..46b0589 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -843,7 +843,7 @@ static int setup_wav_file(int chn) return check_wav_file(chn, given_test_wav_file); #ifdef CONFIG_SUPPORT_CHMAP - if (channel_map_set && chn < channel_map->channels) { + if (channel_map && chn < channel_map->channels) { int channel = channel_map->pos[chn] - SND_CHMAP_FL; if (channel >= 0 && channel < MAX_CHANNELS) return check_wav_file(chn, wavs[channel]); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
by Kuninori Morimoto 07 Dec '16

07 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ssi->usrcnt will be updated on snd_soc_dai_ops::trigger, but snd_pcm_ops::hw_params will be called *before* it. Thus, ssi->usrcnt is still 0 when 1st call. rsnd_ssi_hw_params() needs to check its called count, this means trigger should be if (ssi->usrcnt) instead of if (ssi->usrcnt > 1). Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 932047a..dcb3d0a 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -433,11 +433,14 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, int chan = params_channels(params); /* - * Already working. - * It will happen if SSI has parent/child connection. + * snd_pcm_ops::hw_params will be called *before* + * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0 + * in 1st call. */ - if (ssi->usrcnt > 1) { + if (ssi->usrcnt) { /* + * Already working. + * It will happen if SSI has parent/child connection. * it is error if child <-> parent SSI uses * different channels. */ -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: rsnd: fixup Suspend/Resume clock issue
by Kuninori Morimoto 07 Dec '16

07 Dec '16

Hi Mark These patches fixup Renesas sound's clock stopping issue when Suspend/Resume. These are tested on R-Car Gen3 Salvator-X board Kuninori Morimoto (2): ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting ASoC: rsnd: enable/disable ADG when suspend/resume timing sound/soc/sh/rcar/adg.c | 59 +++++++++++++++++++++++++++++------------------- sound/soc/sh/rcar/core.c | 24 ++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 3 +++ 3 files changed, 63 insertions(+), 23 deletions(-) -- 2.7.4

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[alsa-devel] Cherry trail and Nau 88l24 codec
by Joseph Wang 07 Dec '16

07 Dec '16

Hi all, I've been trying to get sound working on my W9S pipo device. The people at Nuvoton have been extraordinarily helpful, but now the issue is writing the machine driver. My work is at https://github.com/joequant/nau8824 I've gotten the drivers to load, but getting timeouts with the firmware.... [ 102.447840] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 102.447872] intel_sst_acpi 808622A8:00: fw returned err -16 [ 103.471448] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 and then [ 116.820782] intel_sst_acpi 808622A8:00: fw returned err -16 [ 116.938921] intel_sst_acpi 808622A8:00: FW sent error response 0x5000e [ 116.939065] intel_sst_acpi 808622A8:00: FW sent error response 0x5000e [ 116.939579] intel_sst_acpi 808622A8:00: FW sent async error msg: [ 116.939614] (null)00040021 00000000 [ 116.939685] (null)00040021 00000000 [ 116.953260] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 116.953704] intel_sst_acpi 808622A8:00: Alloc for str 3 pipe 0xe [ 116.954062] sst-mfld-sst-mfld-platform: SET_SWM cmd max inputs reached One other path is that ICube has gotten this codec to work on their iwork8 Ultimate android device. They are required to release the source under GPL2, but haven't done so, so if all else fails, it will be a matter of tracking down someone in that company and explaining the issues to them. Since I'm in Hong Kong, I might just find where the things are manufactured, and just show up at their office. But it appears that there are some "magic bits" that are missing. One question that I have is what is the proper DAI to use (i.e. PCM or I2S) for cherry trail. The second question is that looking at the windows driver for nau88l24, there appears to be a test mode that runs the driver against a Realtek rt5645. Should I assume that this means that the initialization is similar? Also I'm quite interested in getting this to work. The W9S Pipo is a perfect netbook that has a big screen that is half the weight of my old laptop. The big thing is that someone realized that instead of putting in a SATA hard disk, that they can use an SD card as main storage, and that cuts down on the weight.

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[alsa-devel] [patch] ALSA: cs46xx: add a new line
by Dan Carpenter 07 Dec '16

07 Dec '16

We accidentally deleted a newline so now the "nreallocated++;" statement is hanging out way off to the right of the screen. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 4a0cbd2241d8..aa61615288ff 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -107,7 +107,8 @@ static int shadow_and_reallocate_code (struct snd_cs46xx * chip, u32 * data, u32 dev_dbg(chip->card->dev, "handle_wideop:[2] %05x:%05x addr %04x\n", - hival, loval, address); nreallocated++; + hival, loval, address); + nreallocated++; } /* wide_opcodes[j] == wide_op */ } /* for */ } /* mod_type == 0 ... */

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[alsa-devel] [PATCH v7 0/3] Media Device Allocator API
by Shuah Khan 07 Dec '16

07 Dec '16

Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. Patches 0001 and 0002 are rebased to 4.9-rc7. Patch 0003 for snd-usb-audio is a rebase of the patch that was tested with the original Media Device Allocator patch series. snd-usb-audio patch includes the fixes found during 4.7-rc1 time in the original snd-usb-audio patch. Changes to patches in this series: Changes to patch 0001 since v6: - No changes Changes to patch 0002 since v6: - No changes Changes to patch 0003 since v6: - Addressed Takashi's review comments on patch v6 Depends on: https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0001 since v5: (comments from Mauro and Sakari) - Removed struct device from media_device_instance. mdev.dev is used instead. - Added documentation. Changes to patch 0002: - No changes since patch v2, applies cleanly on top of the following: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0003: - Changed to hold graph_mutex to check and call enable_source and disable_source handlers - to match au0828 doing the same in: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0001 since v4: - Addressed Sakari's review comments with the exception of opting to not introduce media_device_usb_allocate() macro, and to not add a new routine to find media device instance to avoid a one line check. Changes to patch 0001 since v3: - Fixed undefined reference to `__media_device_usb_init compile error when CONFIG_USB is disabled. - Fixed kernel paging error when accessing /dev/mediaX after rmmod of the module that owns the media_device. The fix bumps the reference count for the owner when second driver comes along to share the media_device. If au0828 owns the media_device, then snd_usb_audio will bump the refcount for au0828, so it won't get deleted and vice versa. Changes to patch 0002 since v2: - Updated media_device_delete() to pass in module name. Changes to patch 0003 since the last version in 4.7-rc1: - Included fixes to bugs found during testing. - Updated to use the Media Allocator API. This patch series has been tested with au0828 and snd-usb-audio drivers. Ran bind and unbind loop tests on each driver with mc_nextgen_test and media_device_test app loop tests while checking lsmod and dmesg. Please refer to tools/testing/selftests/media_tests/regression_test.txt for testing done on this series. Shuah Khan (3): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 ++++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 133 ++++++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 54 ++++++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 321 +++++++++++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 17 files changed, 684 insertions(+), 14 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.7.4

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[alsa-devel] [patch] dbri: move dereference after check for NULL
by Dan Carpenter 06 Dec '16

06 Dec '16

We accidentally introduced a dereference before the NULL check in xmit_descs() as part of silencing a GCC warning. Fixes: 16f46050e709 ("dbri: Fix compiler warning") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Goes into the sparc tree. diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 3fe4468..52063b2 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -1702,7 +1702,7 @@ interrupts are disabled. static void xmit_descs(struct snd_dbri *dbri) { struct dbri_streaminfo *info; - u32 dvma_addr = (u32)dbri->dma_dvma; + u32 dvma_addr; s32 *cmd; unsigned long flags; int first_td; @@ -1710,6 +1710,7 @@ static void xmit_descs(struct snd_dbri *dbri) if (dbri == NULL) return; /* Disabled */ + dvma_addr = (u32)dbri->dma_dvma; info = &dbri->stream_info[DBRI_REC]; spin_lock_irqsave(&dbri->lock, flags);

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[alsa-devel] Applied "ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture" to the asoc tree
by Mark Brown 06 Dec '16

06 Dec '16

The patch ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c1aee1d828caa86a663c57a56eead7fd5f22d80f Mon Sep 17 00:00:00 2001 From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Date: Tue, 6 Dec 2016 07:08:36 -0800 Subject: [PATCH] ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture This patch updates FE channel constraints & BE fixup to support quad channel DMIC capture. DMIC pin's BE fixup is configured based on channel input, i.e. either stereo or quad. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bff80b467262..3a8c24a59020 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -30,6 +30,7 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" #define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 static struct snd_soc_jack broxton_headset; @@ -182,6 +183,16 @@ static struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + static int bxt_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -258,7 +269,10 @@ static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = DUAL_CHANNEL; + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -267,9 +281,9 @@ static int broxton_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.channels_max = DUAL_CHANNEL; + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); + &constraints_channels_quad); return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -- 2.10.2

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[alsa-devel] [PATCH] ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
by sathyanarayana.nujella＠intel.com 06 Dec '16

06 Dec '16

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> This patch updates FE channel constraints & BE fixup to support quad channel DMIC capture. DMIC pin's BE fixup is configured based on channel input, i.e. either stereo or quad. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bff80b4..3a8c24a 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -30,6 +30,7 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" #define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 static struct snd_soc_jack broxton_headset; @@ -182,6 +183,16 @@ static struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + static int bxt_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -258,7 +269,10 @@ static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = DUAL_CHANNEL; + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -267,9 +281,9 @@ static int broxton_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.channels_max = DUAL_CHANNEL; + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); + &constraints_channels_quad); return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -- 2.8.4

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[alsa-devel] [PATCH 0/2 v2] usb-audio misc fix
by Jiada Wang 06 Dec '16

06 Dec '16

This patch set contains the following patches Andreas Pape (1): ALSA: usb-audio: more tolerant packetsize Daniel Girnus (1): ALSA: usb-audio: avoid setting of sample rate multiple times on bus sound/usb/endpoint.c | 4 ++-- sound/usb/pcm.c | 21 +++++++++++---------- 2 files changed, 13 insertions(+), 12 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH] ASoC: nau8825: disable sinc filter for high THD of ADC
by John Hsu 06 Dec '16

06 Dec '16

This bit will enable 4th order SINC filter. =1, filter will enable; but it consumes higher power. =0, the sinc filter is disable, and it should always keep 0 value to get high THD. Therefor, disable the filter when codec initiation for better performance when recording. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 3 ++- sound/soc/codecs/nau8825.h | 2 ++ 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 2b89569..efe3a44 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1846,7 +1846,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) * (audible hiss). Set it to something better. */ regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, - NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_64); + NAU8825_ADC_SYNC_DOWN_MASK | NAU8825_ADC_SINC4_EN, + NAU8825_ADC_SYNC_DOWN_64); regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_64); /* Disable DACR/L power */ diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 0672a25..5d1704e 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -269,6 +269,8 @@ #define NAU8825_BIQ_PATH_DAC (1 << NAU8825_BIQ_PATH_SFT) /* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SINC4_SFT 4 +#define NAU8825_ADC_SINC4_EN (1 << NAU8825_ADC_SINC4_SFT) #define NAU8825_ADC_SYNC_DOWN_SFT 0 #define NAU8825_ADC_SYNC_DOWN_MASK 0x3 #define NAU8825_ADC_SYNC_DOWN_32 0 -- 2.6.4

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[alsa-devel] [PATCH] ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
by Axel Lin 06 Dec '16

06 Dec '16

The logic of "value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_XXXXXX;" is unnecessary complex. By setting CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK as the mask for regmap_update_bits() call, what the code does is exactly the same as setting value = CS35L34_MCLK_RATE_XXXXXX. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l34.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index e0f672a..7c5d151 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -610,15 +610,15 @@ static int cs35l34_dai_set_sysclk(struct snd_soc_dai *dai, switch (freq) { case CS35L34_MCLK_5644: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_5P6448; + value = CS35L34_MCLK_RATE_5P6448; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_6: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P0000; + value = CS35L34_MCLK_RATE_6P0000; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_6144: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P1440; + value = CS35L34_MCLK_RATE_6P1440; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_11289: -- 2.9.3

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[alsa-devel] [PATCH] ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
by Takashi Iwai 06 Dec '16

06 Dec '16

HP Z1 Gen3 AiO with Conexant codec doesn't give an unsolicited event to the headset mic pin upon the jack plugging, it reports only to the headphone pin. It results in the missing mic switching. Let's fix up by simply gating the jack event. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ed62748a6d55..c15c51bea26d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -262,6 +262,7 @@ enum { CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, CXT_FIXUP_HP_SPECTRE, + CXT_FIXUP_HP_GATE_MIC, }; /* for hda_fixup_thinkpad_acpi() */ @@ -633,6 +634,17 @@ static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec, (1 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* the mic pin (0x19) doesn't give an unsolicited event; + * probe the mic pin together with the headphone pin (0x16) + */ + if (action == HDA_FIXUP_ACT_PROBE) + snd_hda_jack_set_gating_jack(codec, 0x19, 0x16); +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -774,6 +786,10 @@ static const struct hda_fixup cxt_fixups[] = { { } } }, + [CXT_FIXUP_HP_GATE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_hp_gate_mic_jack, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -824,6 +840,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), + SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.11.0

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[alsa-devel] [PATCH v2] ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
by Kai-Heng Feng 06 Dec '16

06 Dec '16

Commit [64047d7f4912 ALSA: hda - ignore the assoc and seq when comparing pin configurations] intented to ignore both seq and assoc at pin comparing, but it only ignored seq. So that commit may still fail to match pins on some machines. Change the bitmask to also ignore assoc. v2: Use macro to do bit masking. Thanks to Hui Wang for the analysis. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4ad29f8..a03cf68 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -884,6 +884,8 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action) } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); +#define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC)) + static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { @@ -901,7 +903,7 @@ static bool pin_config_match(struct hda_codec *codec, for (; t_pins->nid; t_pins++) { if (t_pins->nid == nid) { found = 1; - if ((t_pins->val & 0xfffffff0) == (cfg & 0xfffffff0)) + if ((t_pins->val & IGNORE_SEQ_ASSOC) == (cfg & IGNORE_SEQ_ASSOC)) break; else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) break; -- 2.10.2

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[alsa-devel] [PATCH] ALSA: hda: ignore assoc when comparing pin configurations
by Kai-Heng Feng 06 Dec '16

06 Dec '16

Commit [64047d7f4912 ALSA: hda - ignore the assoc and seq when comparing pin configurations] may still fail to match pins on some machines, because the bitmask it used only ignore seq but not assoc. Change the bitmask to also ignore assoc. Thanks to Hui Wang for the analysis. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4ad29f8..fdc30d0 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -901,7 +901,7 @@ static bool pin_config_match(struct hda_codec *codec, for (; t_pins->nid; t_pins++) { if (t_pins->nid == nid) { found = 1; - if ((t_pins->val & 0xfffffff0) == (cfg & 0xfffffff0)) + if ((t_pins->val & 0xffffff00) == (cfg & 0xffffff00)) break; else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) break; -- 2.10.2

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[alsa-devel] [PATCH v2 0/3] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Sebastien Guiriec (3): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version sound/soc/intel/atom/sst/sst.c | 39 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++++++- 3 files changed, 49 insertions(+), 1 deletion(-) -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

3 5

[alsa-devel] Applied "ASoC: Intel: atom: save FW version" to the asoc tree
by Mark Brown 05 Dec '16

05 Dec '16

The patch ASoC: Intel: atom: save FW version has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c7da1e9d2641f67e8b20703118ac02fe79604f10 Mon Sep 17 00:00:00 2001 From: Sebastien Guiriec <sebastien.guiriec(a)intel.com> Date: Mon, 5 Dec 2016 17:45:01 +0100 Subject: [PATCH] ASoC: Intel: atom: save FW version After the boot of the SST FW the firmware version is send back to the driver. This patch is saving the FW version inside the driver. Signed-off-by: Sebastien Guiriec <sebastien.guiriec(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 6 ++++++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 3f493862e98d..5c9a51cc77aa 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -436,6 +436,7 @@ struct intel_sst_drv { */ char firmware_name[FW_NAME_SIZE]; + struct snd_sst_fw_version fw_version; struct sst_fw_save *fw_save; }; diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 7934a0047384..374bb61c596d 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -242,6 +242,12 @@ static void process_fw_init(struct intel_sst_drv *sst_drv_ctx, dev_dbg(sst_drv_ctx->dev, "Build date %s Time %s\n", init->build_info.date, init->build_info.time); + /* Save FW version */ + sst_drv_ctx->fw_version.type = init->fw_version.type; + sst_drv_ctx->fw_version.major = init->fw_version.major; + sst_drv_ctx->fw_version.minor = init->fw_version.minor; + sst_drv_ctx->fw_version.build = init->fw_version.build; + ret: sst_wake_up_block(sst_drv_ctx, retval, FW_DWNL_ID, 0 , NULL, 0); } -- 2.10.2

1 0

[alsa-devel] [PATCH v5 0/4] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Update since V3: - Add sysfs entry description. Update since V4: - Add ack from vinod on documentation - Resend on top of v4.9-rc1. Sebastien Guiriec (4): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version Documentation: ABI: Add ASoC Intel SST Atom sysfs description Documentation/ABI/testing/sysfs-platform-sst-atom | 17 ++++++++++ sound/soc/intel/atom/sst/sst.c | 39 +++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++- 4 files changed, 66 insertions(+), 1 deletion(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 6

[alsa-devel] [PATCH v6 0/3] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Update since V3: - Add sysfs entry description. Update since V4: - Add ack from vinod on documentation - Resend on top of v4.9-rc1. Update since V5: - Resend on top of topic/intel - Remove first patch as the first version was already applied inside topic/intel without getting "Applied" notifiaction. https://www.spinics.net/lists/alsa-devel/msg56122.html Sebastien Guiriec (3): ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version Documentation: ABI: Add ASoC Intel SST Atom sysfs description Documentation/ABI/testing/sysfs-platform-sst-atom | 17 ++++++++++ sound/soc/intel/atom/sst/sst.c | 39 +++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 6 ++++ 4 files changed, 63 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 3

Re: [alsa-devel] [PATCH v3] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Peter Ujfalusi 05 Dec '16

05 Dec '16

Hi, On 12/01/2016 03:19 AM, Matt Ranostay wrote: > We can get audio errors if hitting deeper idle states on omaps: > > [alsa.c:230] error: Fatal problem with alsa output, error -5. > [audio.c:614] error: Error in writing audio (Input/output error?)! > > This seems to happen with off mode idle enabled as power for the > whole SoC may get cut off between filling the McBSP fifo using DMA. > While active DMA blocks deeper idle states in hardware, McBSP > activity does not seem to do so. > > Basing the QoS latency on the FIFO size with the approximate delay > required being 2.3 milliseconds per buffer location. > > Note that this is different from snd_pcm_hw_constraint_step() as > that can configure things based on the buffer and period size. > However, that does not help to block idle between the fifo fills. > > Note that in theory some omaps are capable of playing audio while > hitting deeper idle states. If somebody needs that and gets it > working, we can set the latency requirements accordingly. > > Based on the original patch by Tony Lindgren > Link: https://patchwork.kernel.org/patch/9305867/ > > Cc: Tony Lindgren <tony(a)atomide.com> > Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> > --- > Changes from v1: > * add calculations for latency per number of FIFO locations > > Changes from v2: > * add missing mcbsp.h header change > > sound/soc/omap/mcbsp.c | 15 +++++++++++++++ > sound/soc/omap/mcbsp.h | 2 ++ > 2 files changed, 17 insertions(+) > > diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c > index 06fec5699cc8..b4a773ca2bac 100644 > --- a/sound/soc/omap/mcbsp.c > +++ b/sound/soc/omap/mcbsp.c > @@ -25,6 +25,7 @@ > #include <linux/io.h> > #include <linux/slab.h> > #include <linux/pm_runtime.h> > +#include <linux/pm_qos.h> > > #include <linux/platform_data/asoc-ti-mcbsp.h> > > @@ -637,12 +638,21 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) > * Here we start the McBSP, by enabling transmitter, receiver or both. > * If no transmitter or receiver is active prior calling, then sample-rate > * generator and frame sync are started. > + * > + * Also setting of the QoS latency for the FIFO which varies upon the buffer > + * size. Approximately 2.3 milliseconds per FIFO location. > */ > void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) > { > int enable_srg = 0; > + int latency = mcbsp->pdata->buffer_size * 23; I think this is not correct. The McBSP FIFO time depends on the sample rate and on the number of channels the audio is using. With 8KHz mono you have 12 times more time per FIFO element compared to 48KHz stereo. As it has been discussed with Tony we should calculate the QoS latency in hw_params: latency_ms = ((FIFOsize - FIFOthreshold) / channels) * 1000/sampling-rate On OMAP3.McBSP2 for example (44.1KHz, stereo): FIFO threshold 128 - DMA request will be triggered when 128 slots are free in the FIFO - at that point we have still 1152 words in the FIFO. - if the C wakeup latency is longer then what it takes to play out the samples from the FIFO (13.06ms), we will drain the FIFO and got underflow. - in this case the QOS should be set as 13.06ms FIFO threshold 1024 - DMA request will be triggered when 1024 slots are free in the FIFO - at that point we have still 256 words in the FIFO. - if the C wakeup latency is longer then what it takes to play out the samples from the FIFO (2.9ms), we will drain the FIFO and got underflow. - in this case the QOS should be 2.9ms On other McBSPs with 128 word FIFO the required latency is shorter to ensure we don't drain the FIFO. > u16 w; > > + /* Prevent omap hardware from hitting off between fifo fills */ > + if (latency) > + pm_qos_add_request(&mcbsp->pm_qos_req, > + PM_QOS_CPU_DMA_LATENCY, latency); > + > if (mcbsp->st_data) > omap_st_start(mcbsp); > > @@ -731,6 +741,8 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) > > if (mcbsp->st_data) > omap_st_stop(mcbsp); > + > + pm_qos_remove_request(&mcbsp->pm_qos_req); > } > > int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) > @@ -1098,6 +1110,9 @@ int omap_mcbsp_init(struct platform_device *pdev) > > void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) > { > + if (pm_qos_request_active(&mcbsp->pm_qos_req)) > + pm_qos_remove_request(&mcbsp->pm_qos_req); > + > if (mcbsp->pdata->buffer_size) > sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); > > diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h > index 61e93b1c185d..e603f33f4082 100644 > --- a/sound/soc/omap/mcbsp.h > +++ b/sound/soc/omap/mcbsp.h > @@ -325,6 +325,8 @@ struct omap_mcbsp { > unsigned int in_freq; > int clk_div; > int wlen; > + > + struct pm_qos_request pm_qos_req; > }; > > void omap_mcbsp_config(struct omap_mcbsp *mcbsp, > -- Péter

3 4

[alsa-devel] [PATCH] ASoC: cs42l56: Fix misuse of regmap_update_bits
by Florian Vaussard 05 Dec '16

05 Dec '16

Using regmap_update_bits(..., mask, 1) with 'mask' following (1 << k) and k greater than 0 is wrong. Indeed, _regmap_update_bits will perform (mask & 1), which results in 0 if LSB of mask is 0. Thus the call regmap_update_bits(..., mask, 1) is in reality equivalent to regmap_update_bits(..., mask, 0). In such a case, the correct use is regmap_update_bits(..., mask, mask). This driver is performing such a mistake with the CS42L56_AIN*_REF_MASK masks, which equal 0x10, 0x20, 0x40 and 0x80. Fix the driver to make it consistent with the API. Please note that this change is untested, as I do not have this piece of hardware. Testers are welcome! Signed-off-by: Florian Vaussard <florian.vaussard(a)heig-vd.ch> --- sound/soc/codecs/cs42l56.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3e2c046..cb6ca85 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1277,19 +1277,23 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, if (cs42l56->pdata.ain1a_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN1A_REF_MASK, 1); + CS42L56_AIN1A_REF_MASK, + CS42L56_AIN1A_REF_MASK); if (cs42l56->pdata.ain1b_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN1B_REF_MASK, 1); + CS42L56_AIN1B_REF_MASK, + CS42L56_AIN1B_REF_MASK); if (cs42l56->pdata.ain2a_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN2A_REF_MASK, 1); + CS42L56_AIN2A_REF_MASK, + CS42L56_AIN2A_REF_MASK); if (cs42l56->pdata.ain2b_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN2B_REF_MASK, 1); + CS42L56_AIN2B_REF_MASK, + CS42L56_AIN2B_REF_MASK); if (cs42l56->pdata.micbias_lvl) regmap_update_bits(cs42l56->regmap, CS42L56_GAIN_BIAS_CTL, -- 2.5.5

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[alsa-devel] [PATCH 1/2] ASoC: topology: Allow a widget to have multiple enum controls
by mengdong.lin＠linux.intel.com 05 Dec '16

05 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This patch can create multiple enumerated mixer controls for a widget. Previously topology kernel driver assumes a widget can have only one emumerated mixer control. We need to remove this restriction for Broxton. Its firmware modules (widgets) may need multiple enum controls based on the channel and MIC combination. No ABI change is needed. The ABI allows a widget to embed multiple controls. Reported-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dfdc65..63e1a50 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -486,21 +486,24 @@ static void remove_widget(struct snd_soc_component *comp, dobj->ops->widget_unload(comp, dobj); /* - * Dynamic Widgets either have 1 enum kcontrol or 1..N mixers. + * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ if (dobj->widget.kcontrol_enum) { /* enumerated widget mixer */ - struct soc_enum *se = - (struct soc_enum *)w->kcontrols[0]->private_value; + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct soc_enum *se = + (struct soc_enum *)kcontrol->private_value; - snd_ctl_remove(card, w->kcontrols[0]); + snd_ctl_remove(card, kcontrol); - kfree(se->dobj.control.dvalues); - for (i = 0; i < se->items; i++) - kfree(se->dobj.control.dtexts[i]); + kfree(se->dobj.control.dvalues); + for (i = 0; i < se->items; i++) + kfree(se->dobj.control.dtexts[i]); - kfree(se); + kfree(se); + } kfree(w->kcontrol_news); } else { /* non enumerated widget mixer */ @@ -1256,98 +1259,105 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( } static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( - struct soc_tplg *tplg) + struct soc_tplg *tplg, int num_kcontrols) { struct snd_kcontrol_new *kc; struct snd_soc_tplg_enum_control *ec; struct soc_enum *se; - int i, err; - - ec = (struct snd_soc_tplg_enum_control *)tplg->pos; - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); - - /* validate kcontrol */ - if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - return NULL; + int i, j, err; - kc = kzalloc(sizeof(*kc), GFP_KERNEL); + kc = kcalloc(num_kcontrols, sizeof(*kc), GFP_KERNEL); if (kc == NULL) return NULL; - se = kzalloc(sizeof(*se), GFP_KERNEL); - if (se == NULL) - goto err; + for (i = 0; i < num_kcontrols; i++) { + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return NULL; + + se = kzalloc(sizeof(*se), GFP_KERNEL); + if (se == NULL) + goto err; - dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", - ec->hdr.name); + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", + ec->hdr.name); - kc->name = ec->hdr.name; - kc->private_value = (long)se; - kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc->access = ec->hdr.access; + kc[i].name = ec->hdr.name; + kc[i].private_value = (long)se; + kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc[i].access = ec->hdr.access; - /* we only support FL/FR channel mapping atm */ - se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_l = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FR); + /* we only support FL/FR channel mapping atm */ + se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); + se->shift_l = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FL); + se->shift_r = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FR); - se->items = ec->items; - se->mask = ec->mask; - se->dobj.index = tplg->index; + se->items = ec->items; + se->mask = ec->mask; + se->dobj.index = tplg->index; - switch (ec->hdr.ops.info) { - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - err = soc_tplg_denum_create_values(se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create values for %s\n", - ec->hdr.name); + switch (ec->hdr.ops.info) { + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + err = soc_tplg_denum_create_values(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create values for %s\n", + ec->hdr.name); + goto err_se; + } + /* fall through to create texts */ + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + err = soc_tplg_denum_create_texts(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + ec->hdr.name); + goto err_se; + } + break; + default: + dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", + ec->hdr.ops.info, ec->hdr.name); goto err_se; } - /* fall through to create texts */ - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - err = soc_tplg_denum_create_texts(se, ec); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&ec->hdr, &kc[i], tplg); + if (err) { + soc_control_err(tplg, &ec->hdr, ec->hdr.name); + goto err_se; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc[i], + (struct snd_soc_tplg_ctl_hdr *)ec); if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + dev_err(tplg->dev, "ASoC: failed to init %s\n", ec->hdr.name); goto err_se; } - break; - default: - dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", - ec->hdr.ops.info, ec->hdr.name); - goto err_se; - } - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&ec->hdr, kc, tplg); - if (err) { - soc_control_err(tplg, &ec->hdr, ec->hdr.name); - goto err_se; - } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, kc, - (struct snd_soc_tplg_ctl_hdr *)ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - ec->hdr.name); - goto err_se; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.size); } return kc; err_se: - /* free values and texts */ - kfree(se->dobj.control.dvalues); - for (i = 0; i < ec->items; i++) - kfree(se->dobj.control.dtexts[i]); + for (; i >= 0; i--) { + /* free values and texts */ + se = (struct soc_enum *)kc[i].private_value; + kfree(se->dobj.control.dvalues); + for (j = 0; j < ec->items; j++) + kfree(se->dobj.control.dtexts[j]); - kfree(se); + kfree(se); + } err: kfree(kc); @@ -1499,9 +1509,10 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: template.dobj.widget.kcontrol_enum = 1; - template.num_kcontrols = 1; + template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = - soc_tplg_dapm_widget_denum_create(tplg); + soc_tplg_dapm_widget_denum_create(tplg, + template.num_kcontrols); if (!template.kcontrol_news) { ret = -ENOMEM; goto hdr_err; -- 2.5.0

2 1

[alsa-devel] [PATCH 2/2] ASoC: topology: Only free TLV for volume mixers of a widget
by mengdong.lin＠linux.intel.com 05 Dec '16

05 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This patch will check the type of embedded controls for a widget, and only free the TLV of volume mixers. Bytes controls don't have TLV. Just free the private value which is used as struct soc_mixer_control for volume mixers or soc_bytes_ext for bytes controls. No need to cast to these types before freeing it. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index d318fe4..86c884a 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -53,7 +53,7 @@ struct snd_soc_dobj_control { /* dynamic widget object */ struct snd_soc_dobj_widget { - unsigned int kcontrol_enum:1; /* this widget is an enum kcontrol */ + unsigned int kcontrol_type; /* kcontrol type: mixer, enum, bytes */ }; /* generic dynamic object - all dynamic objects belong to this struct */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 63e1a50..11feb19 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -489,7 +489,7 @@ static void remove_widget(struct snd_soc_component *comp, * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ - if (dobj->widget.kcontrol_enum) { + if (dobj->widget.kcontrol_type == SND_SOC_TPLG_TYPE_ENUM) { /* enumerated widget mixer */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; @@ -506,16 +506,21 @@ static void remove_widget(struct snd_soc_component *comp, } kfree(w->kcontrol_news); } else { - /* non enumerated widget mixer */ + /* volume mixer or bytes controls */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; - struct soc_mixer_control *sm = - (struct soc_mixer_control *) kcontrol->private_value; - kfree(w->kcontrols[i]->tlv.p); + if (dobj->widget.kcontrol_type + == SND_SOC_TPLG_TYPE_MIXER) + kfree(kcontrol->tlv.p); - snd_ctl_remove(card, w->kcontrols[i]); - kfree(sm); + snd_ctl_remove(card, kcontrol); + + /* Private value is used as struct soc_mixer_control + * for volume mixers or soc_bytes_ext for bytes + * controls. + */ + kfree((void *)kcontrol->private_value); } kfree(w->kcontrol_news); } @@ -1439,6 +1444,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, struct snd_soc_dapm_widget template, *widget; struct snd_soc_tplg_ctl_hdr *control_hdr; struct snd_soc_card *card = tplg->comp->card; + unsigned int kcontrol_type; int ret = 0; if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == @@ -1494,6 +1500,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_VOLSW_XR_SX: case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: + kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; /* volume mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dmixer_create(tplg, @@ -1508,7 +1515,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - template.dobj.widget.kcontrol_enum = 1; + kcontrol_type = SND_SOC_TPLG_TYPE_ENUM; /* enumerated mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_denum_create(tplg, @@ -1519,6 +1526,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } break; case SND_SOC_TPLG_CTL_BYTES: + kcontrol_type = SND_SOC_TPLG_TYPE_BYTES; /* bytes control */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dbytes_create(tplg, @@ -1555,6 +1563,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } widget->dobj.type = SND_SOC_DOBJ_WIDGET; + widget->dobj.widget.kcontrol_type = kcontrol_type; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; kfree(template.sname); -- 2.5.0

3 2

Re: [alsa-devel] ProjectMix IO Feedback
by Takashi Sakamoto 05 Dec '16

05 Dec '16

\Hi Darren, (C.C.ed to alsa-devel and ffado-user) On Dec 1 2016 19:47, Darren Anderson wrote: > Hi Takashi, > > Just wanted to give you some feedback on how the ProjectMix IO is working > with your changes that were merged into the Linux Kernel. > > Out of the box, on Ubuntu with kernel version 4.4.0-31, on boot, the > interface makes its distinctive popping sound as it sets the clock rate, > showing that the driver is loading fine. It is also now detected by > PulseAudio (without using a jack sink, as I previously had to). I'm then > able to use ffado-mixer to control the device. Thanks for your feedback to ALSA bebob driver. But it's better to post this kind of report to public mailing list, instead of personal address. There might be people who need the information. > There are, however, three issues. > > Firstly, the sample rate seems to be set every time an audio stream is > about to begin. This can be quite irritating at times, due to the > aforementioned distinctive popping sound and lag in the starting of audio > streams. Would it be possible to set the sample rate once at boot time, and > leave it at this unless the user specifically instructs otherwise? Applications can set sampling rate as they want, by ALSA PCM interface. I think it better to keep it what it is unless there's few special cases. For example, ALSA dice driver performs what you expected, due to hardware restriction. Application can't set sampling rate by ALSA PCM interface and they require the other application for this purpose. > Secondly, the main audio seems to be routed from Pulse through the "Aux > 1/2" channel, as opposed to "Mixer 1/2". The device will output no audio > until the "Analog 1/2" output is set to use "Aux 1/2" in FFADO mixer. > However in fixing this issue, a separate issue may be created. As most > distributions play some sort of "start-up sound", and the device seems to > initialize to full volume unless instructed otherwise, the fact that the > output channel needs to be changed may actually save some ear drums. It may > be wise to ask the device to reduce to something reasonable like 25% volume > when the driver is loaded. Or perhaps just mute all outputs by default. Just after booting up, your device has no routing configuration, therefore it generates no sound till configured with use space applications. Default settings of gain/volume are 0dB, perhaps. But this can also be configured from user space applications. What I can do is to produce a way to control devices for applications in user land, and actual way to use the device is left to application developers. If users have something to do, they would do it. > And finally, I've have found that the quality of the audio output from the > device is of a somewhat lower fidelity than the equivalent output on > Windows. I'm currently testing the interface using my KRK Rokit monitor > speakers and finding that the output is lacking in low frequencies. It > sounds as if there is some high pass filtering going on. ALSA bebob driver allows applications to use the device with both of 16 bit sample and 24bit sample with adjusted 8 bit, for playbacking. I guess that your trial is to playback 16 bit PCM sample and looks with highpass filter. Please check the property of PCM substream. > I hope that you find this feedback useful to your ongoing efforts. You're > doing great work. > > If I can do anything else to help, don't hesitate to ask. Cheers Takashi Sakamoto

1 0

[alsa-devel] [PATCH] ASoC: rsnd: rsnd_get_dalign() needs to care SSIU, not SSI
by Kuninori Morimoto 05 Dec '16

05 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSIU was controlled by SSI before, but commit c7f69ab53("ASoC: rsnd: use mod base common method on SSIU") separated it into ssiu.c But, it didn't care about rsnd_get_dalign() for judging SSI_BUSIF_DALIGN register value which changes the stream data order. This function will be called from cmd/src/ssiu now, but current code still cares ssi, not ssiu. This patch fix it up Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 6 +++--- sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f181410..28ef9ab 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -306,7 +306,7 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) */ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 val = 0x76543210; @@ -315,11 +315,11 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) if (rsnd_io_is_play(io)) { struct rsnd_mod *src = rsnd_io_to_mod_src(io); - target = src ? src : ssi; + target = src ? src : ssiu; } else { struct rsnd_mod *cmd = rsnd_io_to_mod_cmd(io); - target = cmd ? cmd : ssi; + target = cmd ? cmd : ssiu; } mask <<= runtime->channels * 4; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a8f61d7..8ca1705 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -382,6 +382,7 @@ struct rsnd_dai_stream { }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) +#define rsnd_io_to_mod_ssiu(io) rsnd_io_to_mod((io), RSND_MOD_SSIU) #define rsnd_io_to_mod_ssip(io) rsnd_io_to_mod((io), RSND_MOD_SSIP) #define rsnd_io_to_mod_src(io) rsnd_io_to_mod((io), RSND_MOD_SRC) #define rsnd_io_to_mod_ctu(io) rsnd_io_to_mod((io), RSND_MOD_CTU) -- 2.7.4

2 1