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

[alsa-devel] Applied "ASoC: Intel: Add DMIC channel constraint for bxt machine" to the asoc tree
by Mark Brown 26 Sep '16

26 Sep '16

The patch ASoC: Intel: Add DMIC channel constraint for bxt machine 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 43c02ede76a60de38504f4f4fd6042f677be093a Mon Sep 17 00:00:00 2001 From: Yong Zhi <yong.zhi(a)intel.com> Date: Mon, 26 Sep 2016 13:02:29 -0700 Subject: [PATCH] ASoC: Intel: Add DMIC channel constraint for bxt machine Add channel and rate constraints for Refcap and dmiccap devices respectively. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 63 +++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3774b117d365..bb4533c305c7 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -37,6 +37,7 @@ enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, BXT_DPCM_AUDIO_REF_CP, + BXT_DPCM_AUDIO_DMIC_CP, BXT_DPCM_AUDIO_HDMI1_PB, BXT_DPCM_AUDIO_HDMI2_PB, BXT_DPCM_AUDIO_HDMI3_PB, @@ -252,6 +253,52 @@ static struct snd_soc_ops broxton_da7219_ops = { .hw_free = broxton_da7219_hw_free, }; +static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels->min = channels->max = DUAL_CHANNEL; + + return 0; +} + +static int broxton_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops broxton_dmic_ops = { + .startup = broxton_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int broxton_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static struct snd_soc_ops broxton_refcap_ops = { + .startup = broxton_refcap_startup, +}; + /* broxton digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broxton_dais[] = { /* Front End DAI links */ @@ -299,6 +346,21 @@ static struct snd_soc_dai_link broxton_dais[] = { .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, + .ops = &broxton_refcap_ops, + }, + [BXT_DPCM_AUDIO_DMIC_CP] + { + .name = "Bxt Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &broxton_dmic_ops, }, [BXT_DPCM_AUDIO_HDMI1_PB] { @@ -382,6 +444,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .codec_dai_name = "dmic-hifi", .platform_name = "0000:00:0e.0", .ignore_suspend = 1, + .be_hw_params_fixup = broxton_dmic_fixup, .dpcm_capture = 1, .no_pcm = 1, }, -- 2.9.3

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[alsa-devel] [PATCH] ASoC: Intel: Add DMIC channel constraint for bxt machine
by Yong Zhi 26 Sep '16

26 Sep '16

Add channel and rate constraints for Refcap and dmiccap devices respectively. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 63 +++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3774b11..bb4533c 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -37,6 +37,7 @@ enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, BXT_DPCM_AUDIO_REF_CP, + BXT_DPCM_AUDIO_DMIC_CP, BXT_DPCM_AUDIO_HDMI1_PB, BXT_DPCM_AUDIO_HDMI2_PB, BXT_DPCM_AUDIO_HDMI3_PB, @@ -252,6 +253,52 @@ static struct snd_soc_ops broxton_da7219_ops = { .hw_free = broxton_da7219_hw_free, }; +static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels->min = channels->max = DUAL_CHANNEL; + + return 0; +} + +static int broxton_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops broxton_dmic_ops = { + .startup = broxton_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int broxton_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static struct snd_soc_ops broxton_refcap_ops = { + .startup = broxton_refcap_startup, +}; + /* broxton digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broxton_dais[] = { /* Front End DAI links */ @@ -299,6 +346,21 @@ static struct snd_soc_dai_link broxton_dais[] = { .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, + .ops = &broxton_refcap_ops, + }, + [BXT_DPCM_AUDIO_DMIC_CP] + { + .name = "Bxt Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &broxton_dmic_ops, }, [BXT_DPCM_AUDIO_HDMI1_PB] { @@ -382,6 +444,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .codec_dai_name = "dmic-hifi", .platform_name = "0000:00:0e.0", .ignore_suspend = 1, + .be_hw_params_fixup = broxton_dmic_fixup, .dpcm_capture = 1, .no_pcm = 1, }, -- 1.9.1

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[alsa-devel] [PATCH v3 0/3] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 26 Sep '16

26 Sep '16

Introduce mechanism to detect sun7i and provide a different regmap different compared to sun4i Allwinner A10. The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because they have different registers. Renamed SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and renamed SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Danny Milosavljevic (3): ASoC: sun4i-codec: rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field name. ASoC: rename "SUN4I_CODEC_SYS_VERI" to "SUN7I_CODEC_AC_DAC_CAL"; rename "SUN4I_CODEC_AC_MIC_PHONE_CAL" to "SUN7I_CODEC_AC_MIC_PHONE_CAL". ASoC: sun4i-codec: Add custom regmap configs for the A10 and A20 variants. sound/soc/sunxi/sun4i-codec.c | 45 ++++++++++++++++++++++++++++++++++--------- 1 file changed, 36 insertions(+), 9 deletions(-)

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[alsa-devel] [PATCH] ASoC: ak4104: Limit the supported sample rates
by Vishal Thanki 26 Sep '16

26 Sep '16

Currently the driver exposes range of sample rates between 8KHz to 192KHz as a part of dai_driver. This does not hold true as the limited number sample rates are allowed in hw_params DAI callback. This patch limits the sample rates exposed via dai_driver. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> --- sound/soc/codecs/ak4104.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index bdca879..1a9d233 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -163,7 +163,10 @@ static struct snd_soc_dai_driver ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- 2.4.11

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[alsa-devel] [PATCH 0/2] ASoC: da7219: Codec soft reset and AAD improvements
by Adam Thomson 26 Sep '16

26 Sep '16

This patch set makes the following updates to the driver code: 1) Ensures codec is properly reset at startup, and if previously active then disable audio paths prior to reset. 2) Disables AAD in suspend, if device is not a wake-up source. Changes are based on top of latest code introduced under the following commits: ASoC: da7219: software reset codec at probe ASoC: da7219: Support HP detect procedure when MCLK not present Adam Thomson (2): ASoC: da7219: Reset codec gracefully, if still active ASoC: da7219: Disable AAD if codec is not a wake-up source include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219-aad.c | 56 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7219-aad.h | 5 ++++ sound/soc/codecs/da7219.c | 60 +++++++++++++++++++++++++++++++------------ sound/soc/codecs/da7219.h | 6 +++++ 5 files changed, 113 insertions(+), 16 deletions(-) -- 1.9.3

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[alsa-devel] Applied "ASoC: dpcm: print dai_link name of BE other than FE." to the asoc tree
by Mark Brown 26 Sep '16

26 Sep '16

The patch ASoC: dpcm: print dai_link name of BE other than FE. 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 94d215cc6fc5f0a5d202ed2b12fa0b14392cc2e4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=E5=BD=AD=E4=B8=9C=E6=9E=97?= <pengdonglin(a)smartisan.com> Date: Mon, 26 Sep 2016 08:29:31 +0000 Subject: [PATCH] ASoC: dpcm: print dai_link name of BE other than FE. When operating the BE, we should print out the dai_link name of BE other than FE. This is useful when analyzing the kernel log. Signed-off-by: Donglin Peng <pengdonglin(a)smartisan.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c1bfe9d8406d..d56a16a0f6fa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1793,7 +1793,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: close BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); soc_pcm_close(be_substream); be_substream->runtime = NULL; @@ -1859,7 +1859,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: hw_free BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); soc_pcm_hw_free(be_substream); @@ -1937,7 +1937,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: hw_params BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); if (ret < 0) { @@ -2017,7 +2017,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, int ret; dev_dbg(dpcm->be->dev, "ASoC: trigger BE %s cmd %d\n", - dpcm->fe->dai_link->name, cmd); + dpcm->be->dai_link->name, cmd); ret = soc_pcm_trigger(substream, cmd); if (ret < 0) @@ -2232,7 +2232,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); ret = soc_pcm_prepare(be_substream); if (ret < 0) { -- 2.9.3

1 0

[alsa-devel] [PATCH v6 0/3] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 26 Sep '16

26 Sep '16

Introduce mechanism to detect sun7i and provide a different regmap (different compared to sun4i Allwinner A10). The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because they have different registers. Rename SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and rename SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Danny Milosavljevic (3): ASoC: sun4i-codec: Rename sun4i_codec_widgets for consistency ASoC: sun4i-codec: Rename some sun7i-only registers ASoC: sun4i-codec: Add custom regmap configs sound/soc/sunxi/sun4i-codec.c | 50 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 41 insertions(+), 9 deletions(-)

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[alsa-devel] [PATCH] ALSA: control: cage TLV_DB_RANGE_HEAD in kernel land because it was obsoleted
by Takashi Sakamoto 26 Sep '16

26 Sep '16

In commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()"), the new macro was added so that "dB range information can be specified without having to count the items manually for TLV_DB_RANGE_HEAD()". In short, TLV_DB_RANGE_HEAD macro was obsoleted. In commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications"), TLV-related macros are exposed for applications in user land to get content of data structured by Type/Length/Value shape. The commit managed to expose TLV-related macros as many as possible, while obsoleted TLV_DB_RANGE_HEAD() was included to the list of exposed macros. This situation brings some confusions to application developers because they might think all exposed macros have their own purpose and useful for applications. For the reason, this commit moves TLV_DB_RANGE_HEAD macro from UAPI header to a header for kernel land, again. The above commit is done within the same development period for kernel 4.9, thus not published yet. This commit might certainly brings no confusions to user land. Reference: commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()") Reference: commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/tlv.h | 9 ++++++++- include/uapi/sound/tlv.h | 3 --- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 6e2e773..3677ebb 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -46,8 +46,15 @@ #define TLV_DB_RANGE_ITEM SNDRV_CTL_TLVD_DB_RANGE_ITEM #define DECLARE_TLV_DB_RANGE SNDRV_CTL_TLVD_DECLARE_DB_RANGE -#define TLV_DB_RANGE_HEAD SNDRV_CTL_TLVD_DB_RANGE_HEAD #define TLV_DB_GAIN_MUTE SNDRV_CTL_TLVD_DB_GAIN_MUTE +/* + * The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR. + * This is an old fasion and obsoleted by commit bf1d1c9b6179("ALSA: tlv: add + * DECLARE_TLV_DB_RANGE()"). + */ +#define TLV_DB_RANGE_HEAD(num) \ + SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) + #endif /* __SOUND_TLV_H */ diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index f3c198f..b4df440 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -94,9 +94,6 @@ unsigned int name[] = { \ SNDRV_CTL_TLVD_DB_RANGE_ITEM(__VA_ARGS__) \ } -/* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */ -#define SNDRV_CTL_TLVD_DB_RANGE_HEAD(num) \ - SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) #define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: rk3399_gru_sound: fix recording pop at first attempt
by Xing Zheng 26 Sep '16

26 Sep '16

From: Wonjoon Lee <woojoo.lee(a)samsung.com> Pop happens when mclk applied but dmic's own boot-time Specify dmic delay times in dt to make sure clocks are ready earlier than dmic working Signed-off-by: Wonjoon Lee <woojoo.lee(a)samsung.com> Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v2: - rename dmic-delay to dmic-wakeup-delay-ms .../bindings/sound/rockchip,rk3399-gru-sound.txt | 7 +++++++ sound/soc/rockchip/rk3399_gru_sound.c | 14 ++++++++++++++ 2 files changed, 21 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt index f19b6c8..eac91db 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -6,10 +6,17 @@ Required properties: connected to the codecs - rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs +Optional properties: +- dmic-wakeup-delay-ms : specify delay time (ms) for DMIC ready. + If this option is specified, which means it's required dmic need + delay for DMIC to ready so that rt5514 can avoid recording before + DMIC send valid data + Example: sound { compatible = "rockchip,rk3399-gru-sound"; rockchip,cpu = <&i2s0>; rockchip,codec = <&max98357a &rt5514 &da7219>; + dmic-wakeup-delay-ms = <20>; }; diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index ee06489..9ed735a 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -38,6 +38,8 @@ #define SOUND_FS 256 +unsigned int rt5514_dmic_delay; + static struct snd_soc_jack rockchip_sound_jack; static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { @@ -123,6 +125,9 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Wait for DMIC stable */ + msleep(rt5514_dmic_delay); + return 0; } @@ -343,6 +348,15 @@ static int rockchip_sound_probe(struct platform_device *pdev) return -ENODEV; } + /* Set DMIC delay */ + ret = device_property_read_u32(&pdev->dev, "dmic-delay", + &rt5514_dmic_delay); + if (ret) { + rt5514_dmic_delay = 0; + dev_dbg(&pdev->dev, + "no optional property 'dmic-delay' found, default: no delay\n"); + } + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].cpu_dai_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL); -- 1.9.1

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[alsa-devel] Applied "ASoC: hdac_hdmi: Drop use of audio component framework to read ELD" to the asoc tree
by Mark Brown 26 Sep '16

26 Sep '16

The patch ASoC: hdac_hdmi: Drop use of audio component framework to read ELD 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 88b456b0f4a2a4c10c141a7d83113a8ce2164463 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Mon, 26 Sep 2016 08:59:50 -0700 Subject: [PATCH] ASoC: hdac_hdmi: Drop use of audio component framework to read ELD The audio component framework code has not yet landed in the i915 driver so drop the use of the API for the time being. Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: Jeeja KP <jeeja.kp(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 201 +++++++++++++++++++++++++++++++------------ 1 file changed, 145 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 537f61aa27fa..4e181b270d95 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -46,10 +46,6 @@ #define ELD_MAX_SIZE 256 #define ELD_FIXED_BYTES 20 -#define ELD_VER_CEA_861D 2 -#define ELD_VER_PARTIAL 31 -#define ELD_MAX_MNL 16 - struct hdac_hdmi_cvt_params { unsigned int channels_min; unsigned int channels_max; @@ -85,6 +81,8 @@ struct hdac_hdmi_pin { hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; struct hdac_hdmi_eld eld; struct hdac_ext_device *edev; + int repoll_count; + struct delayed_work work; struct mutex lock; bool chmap_set; unsigned char chmap[8]; /* ALSA API channel-map */ @@ -175,6 +173,80 @@ static int hdac_hdmi_eld_limit_formats(struct snd_pcm_runtime *runtime, } + /* HDMI ELD routines */ +static unsigned int hdac_hdmi_get_eld_data(struct hdac_device *codec, + hda_nid_t nid, int byte_index) +{ + unsigned int val; + + val = snd_hdac_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, + byte_index); + + dev_dbg(&codec->dev, "HDMI: ELD data byte %d: 0x%x\n", + byte_index, val); + + return val; +} + +static int hdac_hdmi_get_eld_size(struct hdac_device *codec, hda_nid_t nid) +{ + return snd_hdac_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, + AC_DIPSIZE_ELD_BUF); +} + +/* + * This function queries the ELD size and ELD data and fills in the buffer + * passed by user + */ +static int hdac_hdmi_get_eld(struct hdac_device *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size) +{ + int i, size, ret = 0; + + /* + * ELD size is initialized to zero in caller function. If no errors and + * ELD is valid, actual eld_size is assigned. + */ + + size = hdac_hdmi_get_eld_size(codec, nid); + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) { + dev_err(&codec->dev, "HDMI: invalid ELD buf size %d\n", size); + return -ERANGE; + } + + /* set ELD buffer */ + for (i = 0; i < size; i++) { + unsigned int val = hdac_hdmi_get_eld_data(codec, nid, i); + /* + * Graphics driver might be writing to ELD buffer right now. + * Just abort. The caller will repoll after a while. + */ + if (!(val & AC_ELDD_ELD_VALID)) { + dev_err(&codec->dev, + "HDMI: invalid ELD data byte %d\n", i); + ret = -EINVAL; + goto error; + } + val &= AC_ELDD_ELD_DATA; + /* + * The first byte cannot be zero. This can happen on some DVI + * connections. Some Intel chips may also need some 250ms delay + * to return non-zero ELD data, even when the graphics driver + * correctly writes ELD content before setting ELD_valid bit. + */ + if (!val && !i) { + dev_err(&codec->dev, "HDMI: 0 ELD data\n"); + ret = -EINVAL; + goto error; + } + buf[i] = val; + } + + *eld_size = size; +error: + return ret; +} + static int hdac_hdmi_setup_stream(struct hdac_ext_device *hdac, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format) @@ -987,59 +1059,32 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) return hdac_hdmi_query_cvt_params(&edev->hdac, cvt); } -static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, +static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev, struct hdac_hdmi_pin *pin) { - unsigned int ver, mnl; - - ver = (pin->eld.eld_buffer[DRM_ELD_VER] & DRM_ELD_VER_MASK) - >> DRM_ELD_VER_SHIFT; - - if (ver != ELD_VER_CEA_861D && ver != ELD_VER_PARTIAL) { - dev_dbg(&edev->hdac.dev, "HDMI: Unknown ELD version %d\n", ver); - return -EINVAL; - } - - mnl = (pin->eld.eld_buffer[DRM_ELD_CEA_EDID_VER_MNL] & - DRM_ELD_MNL_MASK) >> DRM_ELD_MNL_SHIFT; - - if (mnl > ELD_MAX_MNL) { - dev_dbg(&edev->hdac.dev, "HDMI: MNL Invalid %d\n", mnl); - return -EINVAL; - } - pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER]; - - return 0; } -static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin) +static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) { struct hdac_ext_device *edev = pin->edev; struct hdac_hdmi_priv *hdmi = edev->private_data; struct hdac_hdmi_pcm *pcm; - int size; + int val; - mutex_lock(&hdmi->pin_mutex); - pin->eld.monitor_present = false; + pin->repoll_count = repoll; - size = snd_hdac_acomp_get_eld(&edev->hdac, pin->nid, -1, - &pin->eld.monitor_present, pin->eld.eld_buffer, - ELD_MAX_SIZE); + pm_runtime_get_sync(&edev->hdac.dev); + val = snd_hdac_codec_read(&edev->hdac, pin->nid, 0, + AC_VERB_GET_PIN_SENSE, 0); - if (size > 0) { - size = min(size, ELD_MAX_SIZE); - if (hdac_hdmi_parse_eld(edev, pin) < 0) - size = -EINVAL; - } + dev_dbg(&edev->hdac.dev, "Pin sense val %x for pin: %d\n", + val, pin->nid); - if (size > 0) { - pin->eld.eld_valid = true; - pin->eld.eld_size = size; - } else { - pin->eld.eld_valid = false; - pin->eld.eld_size = 0; - } + + mutex_lock(&hdmi->pin_mutex); + pin->eld.monitor_present = !!(val & AC_PINSENSE_PRESENCE); + pin->eld.eld_valid = !!(val & AC_PINSENSE_ELDV); pcm = hdac_hdmi_get_pcm(edev, pin); @@ -1061,23 +1106,66 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin) } mutex_unlock(&hdmi->pin_mutex); - return; + goto put_hdac_device; } if (pin->eld.monitor_present && pin->eld.eld_valid) { - if (pcm) { - dev_dbg(&edev->hdac.dev, - "jack report for pcm=%d\n", - pcm->pcm_id); + /* TODO: use i915 component for reading ELD later */ + if (hdac_hdmi_get_eld(&edev->hdac, pin->nid, + pin->eld.eld_buffer, + &pin->eld.eld_size) == 0) { - snd_jack_report(pcm->jack, SND_JACK_AVOUT); - } + if (pcm) { + dev_dbg(&edev->hdac.dev, + "jack report for pcm=%d\n", + pcm->pcm_id); + + snd_jack_report(pcm->jack, SND_JACK_AVOUT); + } + hdac_hdmi_parse_eld(edev, pin); + + print_hex_dump_debug("ELD: ", + DUMP_PREFIX_OFFSET, 16, 1, + pin->eld.eld_buffer, pin->eld.eld_size, + true); + } else { + pin->eld.monitor_present = false; + pin->eld.eld_valid = false; + + if (pcm) { + dev_dbg(&edev->hdac.dev, + "jack report for pcm=%d\n", + pcm->pcm_id); - print_hex_dump_debug("ELD: ", DUMP_PREFIX_OFFSET, 16, 1, - pin->eld.eld_buffer, pin->eld.eld_size, false); + snd_jack_report(pcm->jack, 0); + } + } } mutex_unlock(&hdmi->pin_mutex); + + /* + * Sometimes the pin_sense may present invalid monitor + * present and eld_valid. If ELD data is not valid, loop few + * more times to get correct pin sense and valid ELD. + */ + if ((!pin->eld.monitor_present || !pin->eld.eld_valid) && repoll) + schedule_delayed_work(&pin->work, msecs_to_jiffies(300)); + +put_hdac_device: + pm_runtime_put_sync(&edev->hdac.dev); +} + +static void hdac_hdmi_repoll_eld(struct work_struct *work) +{ + struct hdac_hdmi_pin *pin = + container_of(to_delayed_work(work), struct hdac_hdmi_pin, work); + + /* picked from legacy HDA driver */ + if (pin->repoll_count++ > 6) + pin->repoll_count = 0; + + hdac_hdmi_present_sense(pin, pin->repoll_count); } static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) @@ -1096,6 +1184,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) pin->edev = edev; mutex_init(&pin->lock); + INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld); return 0; } @@ -1306,7 +1395,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port) list_for_each_entry(pin, &hdmi->pin_list, head) { if (pin->nid == pin_nid) - hdac_hdmi_present_sense(pin); + hdac_hdmi_present_sense(pin, 1); } } @@ -1407,7 +1496,7 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) } list_for_each_entry(pin, &hdmi->pin_list, head) - hdac_hdmi_present_sense(pin); + hdac_hdmi_present_sense(pin, 1); /* Imp: Store the card pointer in hda_codec */ edev->card = dapm->card->snd_card; @@ -1472,7 +1561,7 @@ static void hdmi_codec_complete(struct device *dev) * all pins here. */ list_for_each_entry(pin, &hdmi->pin_list, head) - hdac_hdmi_present_sense(pin); + hdac_hdmi_present_sense(pin, 1); pm_runtime_put_sync(&edev->hdac.dev); } -- 2.9.3

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