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

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Add 96k sample rate
by Jeremy McDermond 19 Apr '16

19 Apr '16

The TLV320AIC32x4 series supports 96ksps rates in hardware. This patch adds the necessary PLL divider values and clock settings to the table to make 96ksps work. Signed-off-by: Jeremy McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6b6d55..2f8480c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -159,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 48k rate */ {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, - {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}, + + /* 96k rate */ + {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1}, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -564,7 +567,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.5.0

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[alsa-devel] [PATCH v2] ASoC: sti: update dt-bindings for tdm mode
by Moise Gergaud 19 Apr '16

19 Apr '16

- Add "TDM" in the st,mode property list - st,mode property is also mandatory for reader - add tdm playback dai-link example Signed-off-by: Moise Gergaud <moise.gergaud(a)st.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- I forgot to include this patch into "[PATCH v2] ASoC: sti: unip tdm mode" serie that has been applied. history: - V2: rename patch with a subject line reflecting the style for the subsystem .../devicetree/bindings/sound/st,sti-asoc-card.txt | 44 +++++++++++++++++++--- 1 file changed, 39 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index f546dd9..4d9a83d 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -37,17 +37,18 @@ Required properties: - dai-name: DAI name that describes the IP. + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec and support IEC HDMI formats (player only). + "SPDIF" connected to SPDIF codec and support SPDIF formats (player only). + "PCM" PCM standard mode for I2S or TDM bus. + "TDM" TDM mode for TDM bus. + Required properties ("st,sti-uni-player" compatibility only): - clocks: CPU_DAI IP clock source, listed in the same order than the CPU_DAI properties. - uniperiph-id: internal SOC IP instance ID. - - IP mode: IP working mode depending on associated codec. - "HDMI" connected to HDMI codec IP and IEC HDMI formats. - "SPDIF"connected to SPDIF codec and support SPDIF formats. - "PCM" PCM standard mode for I2S or TDM bus. - Optional properties: - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for external codecs connection. @@ -56,6 +57,22 @@ Optional properties: Example: + sti_uni_player1: sti-uni-player@1 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + st,dai-name = "Uni Player #1 (I2S)"; + dma-names = "tx"; + st,uniperiph-id = <1>; + st,version = <5>; + st,mode = "TDM"; + }; + sti_uni_player2: sti-uni-player@2 { compatible = "st,sti-uni-player"; status = "okay"; @@ -99,6 +116,7 @@ Example: dma-names = "rx"; dai-name = "Uni Reader #1 (HDMI RX)"; version = <3>; + st,mode = "PCM"; }; 2) sti-sas-codec: internal audio codec IPs driver @@ -152,4 +170,20 @@ Example of audio card declaration: sound-dai = <&sti_sasg_codec 0>; }; }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; }; -- 1.9.1

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[alsa-devel] [PATCH v4 0/4] Propagate errors out from compressed streams
by Charles Keepax 19 Apr '16

19 Apr '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v3: - Add missing comment for new field in snd_compr_stream - Return -EPIPE on XRUN to match what the PCM framework does It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 76 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 ++++-- sound/soc/soc-compress.c | 5 +-- 4 files changed, 88 insertions(+), 9 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: core: export snd_soc_get_dai_name
by PC Liao 19 Apr '16

19 Apr '16

From: Jean-Francois Moine <moinejf(a)free.fr> snd_soc_get_dai_name() may be used to define a sound card with a different syntax from the one of the simple-card. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- [1] is applied and needs this patch. Otherwise, [1] will build fail. [1] https://patchwork.kernel.org/patch/8878491/ ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 5 +++-- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a21..03b2d19 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1669,6 +1669,8 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, struct device_node **framemaster); +int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name); int snd_soc_of_get_dai_link_codecs(struct device *dev, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..69cbbb0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3786,8 +3786,8 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); -static int snd_soc_get_dai_name(struct of_phandle_args *args, - const char **dai_name) +int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name) { struct snd_soc_component *pos; struct device_node *component_of_node; @@ -3838,6 +3838,7 @@ static int snd_soc_get_dai_name(struct of_phandle_args *args, mutex_unlock(&client_mutex); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_get_dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name) -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] ASoC: da7219: Small clocking related updates for PLL and sysclk
by Adam Thomson 19 Apr '16

19 Apr '16

The patch set contains 2 small fixes/updates relating to clocking in the codec. The first improves PLL locking for a corner scenario with a low frequency MCLK provided, and the second is to remove 32KHz support from the codec set_dai_sysclk() function, to align with updates already made to the set_dai_pll() function. Patches are based on v4.6-rc4 kernel version. Adam Thomson (2): ASoC: da7219: Update PLL ranges and dividers to improve locking ASoC: da7219: Disallow unsupported 32KHz clock setting in set_dai_sysclk() sound/soc/codecs/da7219.c | 30 +++++++++++++++--------------- sound/soc/codecs/da7219.h | 20 ++++++++++---------- 2 files changed, 25 insertions(+), 25 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH v9 0/7] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 19 Apr '16

19 Apr '16

Couple of minor changes since v8 to ALSA/ASoC side patches. The series is based on top of drm-next. However, it does not look like there is enough attention to the DRM side so that the tda998x patches would get in any time soon. So I am happy to rebase the first ALSA-side patches on top of Mark Brown's for-next tree, for instance, if they would have a change to go in on their own. The first two IEC958 helper patches should apply as such. There is currently two other patch series[2][3] that depend on the first three (ALSA-) patches of this series. The third ("ASoC: hdmi-codec: Add audio abort() callback for video side to use") is currently not used and can be dropped if so decided. The rest depends on those the first patches and adds hdmi-audio support for Beaglebone-black. Best regards, Jyri [1] "[PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105524.html [2] "[PATCH v6 00/10] ASoC: Add mediatek HDMI codec support" by Philipp Zabel http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [3] "[RFC v2 0/6] sti: add audio interface to the hdmi driver" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/103374.ht… and "[PATCH v4 0/6] add IEC958 channel status control helpers" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html by Arnaud Pouliquen Changes since v8 * "ALSA: pcm: add IEC958 channel status helper for hw_params" - Split out the 32 bit support - add: "ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper" * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Set hw_params->msbits to 24 in hw_params() callback if sample width is greater than 24 bits Changes since v7 * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Remove the unused "int ret" variable from tda998x_configure_audio() Changes since v6 * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Add "Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com>" and "Tested-by: Philipp Zabel <p.zabel(a)pengutronix.de>", no other changes * "ALSA: pcm: add IEC958 channel status helper for hw_param" - Added kernel-doc for snd_pcm_create_iec958_consumer_hw_params() * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Use PTR_ERR_OR_ZERO() in tda998x_audio_codec_init() - Fix possible uninitialized use of 'size' in tda998x_get_audio_ports() - Store audio configuration in tda998x_audio_hw_params() instead of tda998x_configure_audio(). Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (7): ALSA: pcm: add IEC958 channel status helper for hw_params ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 275 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 65 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 414 +++++++++++++++++++++ 12 files changed, 917 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

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[alsa-devel] [PATCH v5] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 19 Apr '16

19 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. This patch depends on [1], because snd_soc_of_get_dai_name() fix the property name "sound-dai". This patch needs to change name to use, so export snd_soc_get_dai_name() which is included in snd_soc_of_get_dai_name(). [1] https://patchwork.kernel.org/patch/5671961/ ("ASoC: core: export snd_soc_get_dai_name") Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v4: Change name from "sound-dai" to "capture-dai" to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 44 ++++++++++++++++++-- 2 files changed, 47 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..802dd1b 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,17 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional properties: +- capture-dai: audio codec dai on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + capture-dai = <&rt5650 1>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..45dea22 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,26 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_dai_capture = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_dai_capture, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_dai_capture, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_err(card->dev, "Can't get the right codec dai\n"); + return -EINVAL; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +124,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +168,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +196,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + const char *codec_dai_capture; + struct of_phandle_args args; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +220,23 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (device_property_present(&pdev->dev, "capture-dai")) { + ret = of_parse_phandle_with_args(pdev->dev.of_node, + "capture-dai", + "#sound-dai-cells", 0, &args); + if (ret < 0) { + dev_err(&pdev->dev, + "%s capture-dai name fail %d\n", + __func__, ret); + return ret; + } + + ret = snd_soc_get_dai_name(&args, &codec_dai_capture); + mt8173_rt5650_codecs[1].dai_name = codec_dai_capture; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

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[alsa-devel] [PATCH v3] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 19 Apr '16

19 Apr '16

Provide *our* view of what the rules are for the different DAI formats, so that we do not have to trust external interpretations for this crucial bit of interoperability. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- Documentation/sound/alsa/soc/clocking.txt | 146 +++++++++++++++++++++++++++++- 1 file changed, 144 insertions(+), 2 deletions(-) Changed PCM mode A to start left word one BCLK after positive LRC edge. Added link to I2S spec as indicated by Mark Brown. Cheers, Peter diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index b1300162e01c..7eb02f0758e7 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -47,5 +47,147 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. - - +The below diagrams all have BCLK as the first signal, LRC as the second signal +and DATA as the third. Below that is an indication about which DATA bits belong +in what channel. + +A "..." marking as DATA indicates that there may be more bits that are not +shown. Also, all DATA bits marked X may or may not be present. DAI +transmitters must add them should LRC not match the word size exactly and DAI +receivers must be prepared to ignore them. DAI transmitters must insert zeros +for I2S and Left Justified modes, and preferably not drive DATA for the DSP +modes for these extra X bits. + + +I2S + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word starts one BCLK cycle after a negative flank of LRC, and +the right channel word starts one BCLK cycle after a positive flank of LRC. The +words start with the MSB. Receivers must truncate words if more bits per word +are transmitted than they can use, and transmitters must pad words with zeros +if LRC is not matching the word size. Also, see +https://www.sparkfun.com/datasheets/BreakoutBoards/I2SBUS.pdf + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified (aka MSB) + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word starts directly after a positive flank of LRC, and the +right channel word starts directly after a negative flank of LRC. The words +start with the MSB. Receivers must truncate words if more bits per word are +transmitted than they can use, and transmitters must pad words with zeros if +LRC is not matching the word size. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Right Justified (aka LSB) + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word end right before a negative flank of LRC, and the right +channel word ends right before a positive flank of LRC. The words end with LSB. +Transmitters must pad words if LRC is not matching the word size and the +transmitter and the receiver have to agree on the word leghth. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | X |...| X |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +DSP mode A + +LRC should have positive flanks synchronized with a negative flank of BCLK. +The left channel word starts one BCLK after a positive flank of LRC, and the +right channel word starts directly after the left channel word. The words +start with MSB. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | | + +LRC examples for DSP mode A + .---. .---. +---' '-------------------------------------------------------' '----- + .-. .-. +---' '---------------------------------------------------------' '------- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +DSP mode B + +LRC should have positive flanks synchronized with a negative flank of BCLK. +The left channel word starts directly after a positive flank of LRC, and the +right channel word starts directly after the left channel word. The words +start with MSB. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | | + +LRC examples for DSP mode B + .---. .---. +---' '-------------------------------------------------------' '----- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +The above diagrams show normal BCLK and LRC clocking where DAI transmitters +change DATA on the falling edge of BCLK and DAI receivers read DATA on the +rising edge. For inverted BCLK it is naturally the other way around. Inverted +LRC is just that. Note that DSP mode A is compatible with DSP mode B with +inverted LRC, and vice versa. + + +Left Justified, inverted BCLK +-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .- + '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified, inverted LRC + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | -- 2.1.4

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[alsa-devel] [PATCH] Documentation: sti-asoc-card: update tdm mode
by Moise Gergaud 19 Apr '16

19 Apr '16

- Add "TDM" in the st,mode property list - st,mode property is also mandatory for reader - add tdm playback dai-link example Signed-off-by: Moise Gergaud <moise.gergaud(a)st.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- I forgot to include this patch into "[PATCH v2] ASoC: sti: unip tdm mode" serie that has been applied. This patch is the same than the one previously sent in http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106392.html .../devicetree/bindings/sound/st,sti-asoc-card.txt | 44 +++++++++++++++++++--- 1 file changed, 39 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index f546dd9..4d9a83d 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -37,17 +37,18 @@ Required properties: - dai-name: DAI name that describes the IP. + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec and support IEC HDMI formats (player only). + "SPDIF" connected to SPDIF codec and support SPDIF formats (player only). + "PCM" PCM standard mode for I2S or TDM bus. + "TDM" TDM mode for TDM bus. + Required properties ("st,sti-uni-player" compatibility only): - clocks: CPU_DAI IP clock source, listed in the same order than the CPU_DAI properties. - uniperiph-id: internal SOC IP instance ID. - - IP mode: IP working mode depending on associated codec. - "HDMI" connected to HDMI codec IP and IEC HDMI formats. - "SPDIF"connected to SPDIF codec and support SPDIF formats. - "PCM" PCM standard mode for I2S or TDM bus. - Optional properties: - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for external codecs connection. @@ -56,6 +57,22 @@ Optional properties: Example: + sti_uni_player1: sti-uni-player@1 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + st,dai-name = "Uni Player #1 (I2S)"; + dma-names = "tx"; + st,uniperiph-id = <1>; + st,version = <5>; + st,mode = "TDM"; + }; + sti_uni_player2: sti-uni-player@2 { compatible = "st,sti-uni-player"; status = "okay"; @@ -99,6 +116,7 @@ Example: dma-names = "rx"; dai-name = "Uni Reader #1 (HDMI RX)"; version = <3>; + st,mode = "PCM"; }; 2) sti-sas-codec: internal audio codec IPs driver @@ -152,4 +170,20 @@ Example of audio card declaration: sound-dai = <&sti_sasg_codec 0>; }; }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; }; -- 1.9.1

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[alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Tan, Seng Kai 19 Apr '16

19 Apr '16

Hi Takashi, I am facing similar problem like this issue, hope to get your advice. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-August/054241.html I try to turn on S24_LE support for HDMI Audio using below patch. When I aplay S24_LE file it works fine. The problem happen when I aplay S24_3LE file, I get and extremely low volume. When I aplay S24_LE file, I get an left justified value in I2S, this is an expected result: Front end S24_LE : 0x12345600(in 4bytes, LSB 0 padding) Back end S24_3LE : 0x12345600(in 4bytes, LSB 0 padding) When I aplay S24_3LE file, I get an extremely low output volume. Front end S24_3LE : 0x123456(in 3bytes, No padding) Back end S24_LE : 0x00123456(in 4bytes, MSB 0 padding) Can you advice how to tell the alsa plugin to pad in LSB instead of MSB? Below are the patch I used to turn on S24_LE for HDA . Regards, Seng Kai >From 33f182fe1f6d1e49d0618559d1212b619ec902ce Mon Sep 17 00:00:00 2001 From: "SengKai,Tan" <seng.kai.tan(a)intel.com> Date: Fri, 8 Apr 2016 17:34:51 +0800 Subject: [PATCH] ALSA: hda: S24_LE format support for HDA Audio This patch is to add S24_LE support for HDMI audio and HDA codec. Before this patch, HDMI and HDA codec audio only supported S16_LE and S32_LE. Let's say if user would like to play S24_LE file, it has to use plugin to convert to either S16_LE or S32_LE which will increase in processing and less efficient in utilize the resources. On the other side, let's say connected HDMI monitor support S16_LE and S24_LE. Before this patch is applied audio has to be converted to S16_LE. That will reduce the audio quality. Signed-off-by: Tan, Seng Kai <seng.kai.tan(a)intel.com> --- sound/pci/hda/hda_codec.c | 6 ++++-- sound/pci/hda/hda_eld.c | 6 ++++-- sound/pci/hda/patch_hdmi.c | 3 ++- 3 files changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 57197be..9e737bb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3501,14 +3501,16 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, if (val & AC_SUPPCM_BITS_32) formats |= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE; if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24)) - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (val & AC_SUPPCM_BITS_24) bps = 24; else if (val & AC_SUPPCM_BITS_20) bps = 20; } else if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24| AC_SUPPCM_BITS_32)) { - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (val & AC_SUPPCM_BITS_32) bps = 32; else if (val & AC_SUPPCM_BITS_24) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 0e6d753..65bfe10 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -613,12 +613,14 @@ void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { if (a->sample_bits & AC_SUPPCM_BITS_20) { - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (maxbps < 20) maxbps = 20; } if (a->sample_bits & AC_SUPPCM_BITS_24) { - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (maxbps < 24) maxbps = 24; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d02eccd..8cbde13 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2489,7 +2489,8 @@ static const struct hda_verb nvhdmi_basic_init_7x_8ch[] = { SNDRV_PCM_RATE_192000) #define SUPPORTED_MAXBPS 24 #define SUPPORTED_FORMATS \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) #endif static int nvhdmi_7x_init_2ch(struct hda_codec *codec) -- 1.9.1

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