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

[alsa-devel] [PATCH v2] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 18 Apr '16

18 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 | 145 +++++++++++++++++++++++++++++- 1 file changed, 143 insertions(+), 2 deletions(-) Now with some words on each format as well. Cheers, Peter diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index b1300162e01c..e74e91bbee94 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -47,5 +47,146 @@ 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. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |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. +Receivers must truncate words if more bits per word are transmitted than they +can use, and transmitters must pad words if LRC is not matching the word size. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | X |...| X |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +DSP mode A + +LRC should have negative flanks synchronized with a negative flank of BCLK. +The left channel word starts directly after a negative 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 1/2] ASoC: arizona: Free speaker thermal IRQs in CODEC remove
by Charles Keepax 18 Apr '16

18 Apr '16

The thermal warning IRQs for the speaker are requested in CODEC probe but never freed. This patch frees them in CODEC remove. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 12 ++++++++++++ sound/soc/codecs/arizona.h | 2 ++ sound/soc/codecs/cs47l24.c | 3 +++ sound/soc/codecs/wm5102.c | 2 ++ sound/soc/codecs/wm5110.c | 2 ++ sound/soc/codecs/wm8997.c | 2 ++ sound/soc/codecs/wm8998.c | 2 ++ 7 files changed, 25 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d8a6823..7b82550 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -251,6 +251,18 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +int arizona_free_spk(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + + arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT_WARN, arizona); + arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT, arizona); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_free_spk); + static const struct snd_soc_dapm_route arizona_mono_routes[] = { { "OUT1R", NULL, "OUT1L" }, { "OUT2R", NULL, "OUT2L" }, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 1ea8e4e..ce0531b 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -307,6 +307,8 @@ extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); extern int arizona_init_mono(struct snd_soc_codec *codec); +extern int arizona_free_spk(struct snd_soc_codec *codec); + extern int arizona_init_dai(struct arizona_priv *priv, int dai); int arizona_set_output_mode(struct snd_soc_codec *codec, int output, diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 6b8b557..c10f71e 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1140,6 +1140,9 @@ static int cs47l24_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + + arizona_free_spk(codec); + return 0; } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a8b3e3f..59fd3e8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1960,6 +1960,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; + arizona_free_spk(codec); + return 0; } diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 83ba70f..2728ac5 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2298,6 +2298,8 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec) arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + arizona_free_spk(codec); + return 0; } diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 52d766e..6b0785b 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1072,6 +1072,8 @@ static int wm8997_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; + arizona_free_spk(codec); + return 0; } diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 0123960..449f666 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1324,6 +1324,8 @@ static int wm8998_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; + arizona_free_spk(codec); + return 0; } -- 2.1.4

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[alsa-devel] [PATCH 0/4] ASoC: sti: tdm mode
by Moise Gergaud 18 Apr '16

18 Apr '16

This patchset enables tdm mode for ASoC sti. Moise Gergaud (4): ASoC: sti: correct typo errors ASoC: sti: tdm: uni tdm mode ASoC: sti: select player for I2S/TDM TX bus Documentation: sti-asoc-card: update tdm mode .../devicetree/bindings/sound/st,sti-asoc-card.txt | 48 ++++- sound/soc/sti/sti_uniperif.c | 144 ++++++++++++- sound/soc/sti/uniperif.h | 220 ++++++++++++++++++-- sound/soc/sti/uniperif_player.c | 182 +++++++++++----- sound/soc/sti/uniperif_reader.c | 229 +++++++++++++++------ 5 files changed, 687 insertions(+), 136 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 18 Apr '16

18 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v2: Use sound-dai to determine the capture path. --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 42 ++++++++++++++++++-- 2 files changed, 44 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..4ed768e 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: +- sound-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>; + sound-dai = <&rt5650 1>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..9f56df9 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_CLK_SEL_I2S1_ASRC); + 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 +196,7 @@ 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; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +219,20 @@ 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, "sound-dai")) { + ret = snd_soc_of_get_dai_name(pdev->dev.of_node, + &codec_dai_capture); + if (ret < 0) { + dev_err(&pdev->dev, + "%s snd_soc_of_get_dai_name fail %d\n", + __func__, ret); + return ret; + } + 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] ALSA: hda/realtek - Add ALC3234 headset mode for Optiplex 9020m
by Bastien Nocera 18 Apr '16

18 Apr '16

The Optiplex 9020m with Haswell-DT processor needs a quirk for the headset jack at the front of the machine to be able to use microphones. A quirk for this model was originally added in 3127899, but c77900e removed it in favour of a more generic version. Unfortunately, pin configurations can changed based on firmware/BIOS versions, and the generic version doesn't have any effect on newer versions of the machine/firmware anymore. With help from David Henningsson <diwic(a)ubuntu.com> Signed-off-by: Bastien Nocera <hadess(a)hadess.net> Tested-by: Bastien Nocera <hadess(a)hadess.net> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1402ba9..810bcee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5449,6 +5449,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), + SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.7.3

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[alsa-devel] [PATCH] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 18 Apr '16

18 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 | 110 +++++++++++++++++++++++++++++- 1 file changed, 108 insertions(+), 2 deletions(-) Now, the reason for me writing this is that I once again had to go hunting the internet to find the rules for these things. And like last time, I found no definitive source and sometimes strange things as well. So, I don't know if the above is 100% good, but it is my understanding of the subject. Please enlighten me on any errors. Cheers, Peter diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index b1300162e01c..8b498d610e10 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -47,5 +47,111 @@ 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, Left and Right Justified modes, and preferably not drive DATA +for the DSP modes for these extra X bits. + + +I2S + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified (aka MSB) + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Right Justified (aka LSB) + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | X |...| X |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +DSP mode A + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + -. -. + '- '- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | | + +LRC examples for DSP mode A + .---. .---. +---' '-------------------------------------------------------' '----- + .-. .-. +-----' '---------------------------------------------------------' '----- +-------. .---------------------------------------------------------. .--- + '-' '-' + + +DSP mode B + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |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 v2 0/3] Add support simple-card parse jack detection via external codec
by Xing Zheng 18 Apr '16

18 Apr '16

Hi, In most cases, many codecs already supports jack detection, previouslly, we need to create a customized machine driver every time. In my opinion, the codec fill the detect_jack explicitly and tell sound framework it supports the jack detection, the simple-card is able to parse the dts file and call the export function to initialize the jace detection via the codec. I think that it will bring better flexibility. Thanks. Changes in v1: - clean up the commit message and notes Xing Zheng (3): ASoC: jack: Add an export of a function that calls the codec jack detection ASoC: simple-card: Add support call the codec jack detection after parse dts ASoC: da7219: Add detect_jack callback in the snd_soc_codec_driver .../devicetree/bindings/sound/simple-card.txt | 17 +++ include/sound/soc.h | 7 ++ sound/soc/codecs/da7219.c | 2 + sound/soc/generic/simple-card.c | 126 +++++++++++++++++++- sound/soc/soc-jack.c | 17 +++ 5 files changed, 166 insertions(+), 3 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA - hda: hdmi check NULL pointer in hdmi_set_chmap
by libin.yang＠linux.intel.com 18 Apr '16

18 Apr '16

From: Libin Yang <libin.yang(a)linux.intel.com> Make sure per_pin is not NULL before using it. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4833c7b..c99289d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1858,6 +1858,8 @@ static void hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx, struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + if (!per_pin) + return; mutex_lock(&per_pin->lock); per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, ARRAY_SIZE(per_pin->chmap)); -- 1.9.1

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[alsa-devel] [PATCH] aplay: fix lurking capture file overwrite bug
by David Fries 18 Apr '16

18 Apr '16

If -d was given to arecord while commit 8aa13eec80eac312e4b99423909387660fb99b8f (now reverted) was in effect, the last read would be shorter than the chunk size, but pcm_read would read and return the chunk size, the samples were discarded, and capture() continued in a loop because count never reached 0. arecord opens a new file each loop iteration, if arecord is dynamically naming files, --use-strftime option or beyond the wave 2GB limit, this will generate a series of header only wave files. If the file is unique the originally recorded data is lost and it will continue overwriting the same file with a header only wave file. While the current pcm_read can't fail (it can exit), it is better to just fix this lurking bug in case it is "fixed" again. --- In my case I wanted to record for 16 hours, I did get the audio, but I also found I had 3.1 million 44 byte files, which took the system 3 hours to unlink, I'm glad I at least discovered this problem the next day.. Debian has the effectively broken pcm_read, after figuring out what was going wrong, and that the current arecord doesn't exhibit the problem I wanted to fix it anyway. There are remaining problems, pcm_read will read chunk_size, but if the requested size is smaller samples are lost, I'm not trying to fix that here. This will at least gracefully close a wave file (update the counts) if read or write returns a size different from requested. aplay/aplay.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 7acaa83..2da7dda 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -3067,11 +3067,14 @@ static void capture(char *orig_name) size_t c = (rest <= (off64_t)chunk_bytes) ? (size_t)rest : chunk_bytes; size_t f = c * 8 / bits_per_frame; - if (pcm_read(audiobuf, f) != f) + if (pcm_read(audiobuf, f) != f) { + in_aborting = 1; break; + } if (write(fd, audiobuf, c) != c) { perror(name); - prg_exit(EXIT_FAILURE); + in_aborting = 1; + break; } count -= c; rest -= c; @@ -3091,7 +3094,7 @@ static void capture(char *orig_name) } if (in_aborting) - break; + prg_exit(EXIT_FAILURE); /* repeat the loop when format is raw without timelimit or * requested counts of data are recorded -- 2.1.4

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[alsa-devel] Community MC13783 driver for Asynchronous DAI mode
by Edison 18 Apr '16

18 Apr '16

HI Community, I'm working on kernel upgrading from FSL BSP to community BSP for my customized IMX27 based board, everything works fine except problem when porting MC13783 audio device driver. On my board, the hardware schematic for audio part is the same as IMX27 ADS, in which MC13783 audio is connect to IMX27 with two SSI (SSI1/SS2). In the FSL 2.6.22 BSP, the design is the same as the hardware schematic, the ALSA driver use SS1 / SS2 as codec and DAC respectively. In the community BSP (4.X), I tried to assign two difference SSI port for MC13783 codec as below /* MC13783 */ static struct mc13xxx_codec_platform_data mx27_ads_codec = { .dac_ssi_port = MC13783_SSI2_PORT, .adc_ssi_port = MC13783_SSI1_PORT, }; In the community BSP, mc13783 audio driver will be bind to asynchronous MC13783 DAI for separated routing for AD/DA (mc13783.c), one called mc13783- hifi-playback and the other called mc13783-hifi-capture, but DAI bind will be certainly unsuccessful because in mc13783 sound soc driver (imx-mc13783.c), the dai_link name is fixed to "mc13783_hifi", so looks like the bind can only be successfully if we use synchronized mode (AD/DA use the same SSI port rather than different ports). That seems to be a big difference between community BSP and FSL BSP. My question: 1. Has someone used the synchronized DAI in their design with community kernel version? Is it works? (I cannot find any reference for mc13783-hifi- playback/mc13783-hifi-capture in the kernel source code)? 2. Can I use synchronized DAI (force ADC/DAC use same SSI) instead of asynchronous DAI on the community version to make audio works without hardware change? (Unfortunately I have done some experiments but seems not work). Regards, Edison

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