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[alsa-devel] [PATCH 1/1] alsabat: add terminate status check for capture thread
by han.lu＠intel.com 18 Apr '16

18 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> In loopback test, alsabat use pthread_join(pthread_t thread, **retval) to wait for the capture thread to terminate. If the capture thread was canceled, PTHREAD_CANCELED is placed in *retval, and the access to the **retval will fail. Add status check to prevent illegal access to the **retval. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/bat.c b/bat/bat.c index e824065..1afdcb4 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -192,6 +192,12 @@ static void test_loopback(struct bat *bat) exit(EXIT_FAILURE); } + /* check if capture thread is canceled or not */ + if (thread_result_capture == PTHREAD_CANCELED) { + fprintf(bat->log, _("Capture canceled.\n")); + return; + } + /* check capture status */ if (*thread_result_capture != 0) { fprintf(bat->err, _("Exit capture thread fail: %d\n"), -- 2.5.0

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[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

2 1

[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

1 3

[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

2 1

[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

3 3

[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

1 0

[alsa-devel] [PATCH 1/2] amidi: ignore not only Active Sensing but all System Real-Time messages
by Clemens Ladisch 17 Apr '16

17 Apr '16

By default, amidi ignores Active Sensing messages because they are sent by many devices in the background and would only interfere with the actual messages that amidi is supposed to capture. However, there are also devices that send Clock messages with the same problem, so it is a better idea to filter out all System Real-Time messages. Reported-by: Martin Tarenskeen <m.tarenskeen(a)zonnet.nl> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- amidi/amidi.1 | 19 +++++++++++++------ amidi/amidi.c | 11 ++++++----- 2 files changed, 19 insertions(+), 11 deletions(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 1b4cfb1..46e9c50 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -1,4 +1,4 @@ -.TH AMIDI 1 "26 Jun 2006" +.TH AMIDI 1 "26 Mar 2016" .SH NAME amidi \- read from and write to ALSA RawMIDI ports @@ -80,7 +80,7 @@ to record a Standard MIDI (.mid) file, use .B arecordmidi(1). .B amidi -will filter out any Active Sensing bytes (FEh), unless the +will filter out System Real-Time messages (F8h...FFh), unless the .I \-a option has been given. @@ -91,7 +91,7 @@ Sends the bytes specified as hexadecimal numbers to the MIDI port. .TP .I \-d, \-\-dump Prints data received from the MIDI port as hexadecimal bytes. -Active Sensing bytes (FEh) will not be shown, unless the +System Real-Time messages (F8h...FFh) will not be shown, unless the .I \-a option has been given. @@ -107,9 +107,16 @@ If this option has not been given, you must press Ctrl+C (or kill to stop receiving data. .TP -.I \-a, \-\-active\-sensing -Does not ignore Active Sensing bytes (FEh) when saving or printing -received MIDI commands. +.I \-a, \-\-all\-messages +Includes System Real-Time messages (F8h...FFh) when saving or +printing received MIDI commands. +By default, these messages are ignored. + +(Clock (F8h) or Active Sensing (FEh) messages are sent in the +background by many devices, but typically are not wanted and would +only clutter up +.B amidi\fR's +output.) .SH EXAMPLES diff --git a/amidi/amidi.c b/amidi/amidi.c index cedf18c..4978249 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -77,7 +77,7 @@ static void usage(void) "-d, --dump print received data as hexadecimal bytes\n" "-t, --timeout=seconds exits when no data has been received\n" " for the specified duration\n" - "-a, --active-sensing don't ignore active sensing bytes\n"); + "-a, --all-messages include system real-time messages\n"); } static void version(void) @@ -418,11 +418,12 @@ int main(int argc, char *argv[]) {"send-hex", 2, NULL, 'S'}, {"dump", 0, NULL, 'd'}, {"timeout", 1, NULL, 't'}, - {"active-sensing", 0, NULL, 'a'}, + {"all-messages", 0, NULL, 'a'}, + {"active-sensing", 0, NULL, 'a'}, /* for backwards compatibility */ { } }; int c, err, ok = 0; - int ignore_active_sensing = 1; + int ignore_system_realtime = 1; int do_send_hex = 0; while ((c = getopt_long(argc, argv, short_options, @@ -461,7 +462,7 @@ int main(int argc, char *argv[]) timeout = atoi(optarg); break; case 'a': - ignore_active_sensing = 0; + ignore_system_realtime = 0; break; default: error("Try `amidi --help' for more information."); @@ -589,7 +590,7 @@ int main(int argc, char *argv[]) } length = 0; for (i = 0; i < err; ++i) - if (!ignore_active_sensing || buf[i] != 0xfe) + if (!ignore_system_realtime || buf[i] < 0xf8) buf[length++] = buf[i]; if (length == 0) continue;

3 10

[alsa-devel] [PATCH] ALSA: hda - Don't trust the reported actual power state
by Takashi Iwai 17 Apr '16

17 Apr '16

We've got a regression report that the recording on Mac with a cirrus codec doesn't work any longer. This turned out to be the missing power up to D0 by power_save_node enablement. After analyzing the traces, we found out that the culprit is that the codec advertises the "actual" power state of a few nodes to be D0 while the "target" power state is D3. This inconsistency is usually OK, as it implies the power transition. But in the case of cirrus codec, this seems to be stuck to D3 while it's not actually D0. This patch addresses the issue by checking the power state difference more strictly. It sends the power-state change verb unless both the target and the actual power states show the given value. We may introduce yet another flag indicating the possible broken hardware power state, but it's anyway safer to set the proper power state even in a transition (at least it's harmless as long as the target state is same). So this simpler change was applied now. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116171 Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ca5b89f088a..dfaf1a93fb8a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -826,7 +826,7 @@ static hda_nid_t path_power_update(struct hda_codec *codec, bool allow_powerdown) { hda_nid_t nid, changed = 0; - int i, state; + int i, state, power; for (i = 0; i < path->depth; i++) { nid = path->path[i]; @@ -838,7 +838,9 @@ static hda_nid_t path_power_update(struct hda_codec *codec, state = AC_PWRST_D0; else state = AC_PWRST_D3; - if (!snd_hda_check_power_state(codec, nid, state)) { + power = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + if (power != (state | (state << 4))) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, state); changed = nid; -- 2.8.1

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Add SPI support
by Jeremy McDermond 16 Apr '16

16 Apr '16

The TLV320AIC32x4 support the control channel over either SPI or I2C. Added the proper interfacing with regmap to make the driver able to handle either possibility. Signed-off-by: Jeremy McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/Kconfig | 5 +- sound/soc/codecs/tlv320aic32x4.c | 181 +++++++++++++++++++++++++++++++++++---- 2 files changed, 165 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 907c804..1ddf7b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,7 +129,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4 if SND_SOC_I2C_AND_SPI select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -768,7 +768,8 @@ config SND_SOC_TLV320AIC31XX select REGMAP_I2C config SND_SOC_TLV320AIC32X4 - tristate + tristate "Texas Instruments TLV320AIC32x4 CODECs" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_TLV320AIC3X tristate "Texas Instruments TLV320AIC3x CODECs" diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f2d3191..1c888b7 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -31,6 +31,7 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/cdev.h> #include <linux/slab.h> #include <linux/clk.h> @@ -287,15 +288,6 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { }, }; -static const struct regmap_config aic32x4_regmap = { - .reg_bits = 8, - .val_bits = 8, - - .max_register = AIC32X4_RMICPGAVOL, - .ranges = aic32x4_regmap_pages, - .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), -}; - static inline int aic32x4_get_divs(int mclk, int rate) { int i; @@ -668,7 +660,7 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { }; static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4, - struct device_node *np) + struct device_node *np, struct device *dev) { aic32x4->swapdacs = false; aic32x4->micpga_routing = 0; @@ -777,6 +769,23 @@ error_ldo: return ret; } +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static const struct regmap_config aic32x4_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AIC32X4_RMICPGAVOL, + .ranges = aic32x4_regmap_pages, + .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), +}; + static int aic32x4_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -790,7 +799,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, if (aic32x4 == NULL) return -ENOMEM; - aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); + aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_i2c_regmap); if (IS_ERR(aic32x4->regmap)) return PTR_ERR(aic32x4->regmap); @@ -802,7 +811,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->micpga_routing = pdata->micpga_routing; aic32x4->rstn_gpio = pdata->rstn_gpio; } else if (np) { - ret = aic32x4_parse_dt(aic32x4, np); + ret = aic32x4_parse_dt(aic32x4, np, &i2c->dev); if (ret) { dev_err(&i2c->dev, "Failed to parse DT node\n"); return ret; @@ -862,12 +871,6 @@ static const struct i2c_device_id aic32x4_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); -static const struct of_device_id aic32x4_of_id[] = { - { .compatible = "ti,tlv320aic32x4", }, - { /* senitel */ } -}; -MODULE_DEVICE_TABLE(of, aic32x4_of_id); - static struct i2c_driver aic32x4_i2c_driver = { .driver = { .name = "tlv320aic32x4", @@ -877,8 +880,148 @@ static struct i2c_driver aic32x4_i2c_driver = { .remove = aic32x4_i2c_remove, .id_table = aic32x4_i2c_id, }; +#endif -module_i2c_driver(aic32x4_i2c_driver); +#if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config aic32x4_spi_regmap = { + .reg_bits = 7, + .pad_bits = 1, + .val_bits = 8, + .read_flag_mask = 0x01, + + .max_register = AIC32X4_RMICPGAVOL, + .ranges = aic32x4_regmap_pages, + .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), +}; + +static int aic32x4_spi_probe(struct spi_device *spi) +{ + struct aic32x4_pdata *pdata = spi->dev.platform_data; + struct aic32x4_priv *aic32x4; + struct device_node *np = spi->dev.of_node; + int ret; + + aic32x4 = devm_kzalloc(&spi->dev, sizeof(struct aic32x4_priv), + GFP_KERNEL); + if (aic32x4 == NULL) + return -ENOMEM; + + aic32x4->regmap = devm_regmap_init_spi(spi, &aic32x4_spi_regmap); + if (IS_ERR(aic32x4->regmap)) + return PTR_ERR(aic32x4->regmap); + + spi_set_drvdata(spi, aic32x4); + + if (pdata) { + aic32x4->power_cfg = pdata->power_cfg; + aic32x4->swapdacs = pdata->swapdacs; + aic32x4->micpga_routing = pdata->micpga_routing; + aic32x4->rstn_gpio = pdata->rstn_gpio; + } else if (np) { + ret = aic32x4_parse_dt(aic32x4, np, &spi->dev); + if (ret) { + dev_err(&spi->dev, "Failed to parse DT node\n"); + return ret; + } + } else { + aic32x4->power_cfg = 0; + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + aic32x4->rstn_gpio = -1; + } + + aic32x4->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(aic32x4->mclk)) { + dev_err(&spi->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); + return PTR_ERR(aic32x4->mclk); + } + + if (gpio_is_valid(aic32x4->rstn_gpio)) { + ret = devm_gpio_request_one(&spi->dev, aic32x4->rstn_gpio, + GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); + if (ret != 0) + return ret; + } + + ret = aic32x4_setup_regulators(&spi->dev, aic32x4); + if (ret) { + dev_err(&spi->dev, "Failed to setup regulators\n"); + return ret; + } + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_aic32x4, &aic32x4_dai, 1); + if (ret) { + dev_err(&spi->dev, "Failed to register codec\n"); + aic32x4_disable_regulators(aic32x4); + return ret; + } + + spi_set_drvdata(spi, aic32x4); + + return 0; +} + +static int aic32x4_spi_remove(struct spi_device *dev) +{ + struct aic32x4_priv *aic32x4 = spi_get_drvdata(dev); + + aic32x4_disable_regulators(aic32x4); + + snd_soc_unregister_codec(&dev->dev); + return 0; +} + +static const struct spi_device_id aic32x4_spi_id[] = { + { "tlv320aic32x4", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, aic32x4_spi_id); + +static struct spi_driver aic32x4_spi_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_spi_probe, + .remove = aic32x4_spi_remove, + .id_table = aic32x4_spi_id, +}; +#endif + +static int __init aic32x4_modinit(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&aic32x4_i2c_driver); + if (ret) + pr_err("Failed to register tlv320aic32x4 I2C driver: %d\n", + ret); +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&aic32x4_spi_driver); + if (ret) + pr_err("Failed to register tlv320aic32x4 SPI driver: %d\n", + ret); +#endif + + return ret; +} +module_init(aic32x4_modinit); + +static void __exit aic32x4_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic32x4_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&aic32x4_spi_driver); +#endif +} +module_exit(aic32x4_exit); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); -- 2.5.0

1 0

[alsa-devel] SPDIF trouble with multiple usb devices
by Scott Doty 15 Apr '16

15 Apr '16

So I'm having the worst luck with optical SPDIF devices. http://www.alsa-project.org/db/?f=d0d48e597f0cdbd1b377aea6f8d73b000301ae8a Recently I hooked up my synthesizer in the configuration that used to work fine, and it didn't work: The Sound Blaster HD usb audio device doesn't seem to be working on modern kernels, nor does the spandy-new SIIG HD Digital 7.1 USB Audio Box. I'll go down the list of devices: _[/proc/asound]_(root@eva)_ # cat cards 0 [Pro ]: USB-Audio - SB X-Fi Surround 5.1 Pro Creative Technology Ltd SB X-Fi Surround 5.1 Pro at usb-0000:00:14.0-4.7.5, ful This device works fine. Unfortunately, it only has optical SPDIF out, no SPDIF input. 1 [Audio ]: USB-Audio - USB2.0 High-Speed True HD Audio CMEDIA USB2.0 High-Speed True HD Audio at usb-0000:00:14.0-1.4, high speed Just got this from Amazon, it is a "SIIG HD Digital 7.1 USB Audio Box". I was hoping the optical SPDIF input would work to replace the Sound Blaster HD below -- but alas, it doesn't. 2 [M8 ]: USB-Audio - MOTIF ES 8 Yamaha MOTIF ES 8 at usb-0000:00:14.0-1.3, full speed This is actually a MIDI-over-USB device, my synth. The output of the synthesizer is optical SPDIF, which _used_ to work with the following device: 3 [HD ]: USB-Audio - USB Sound Blaster HD Creative Technology USB Sound Blaster HD at usb-0000:00:14.0-4.7.6, full speed I used to be able to use the optical SPDIF input for this device from my synth, but that seems to have stopped working. Further, I can't play audio out of the Sound Blaster HD's optical SPDIF output, either. It works in Windows 10, which is where I had to spend my latest music session -- yuck. Never mind the rest of the devices. Naturally, I'd love for this to be a quick fix, but I'm even willing to buy more hardware, if I know the SPDIF optical input works. (Buying the SIIG device was my way of "punting" by getting Creative out of the mix.) I'm also willing to ship the non-working devices to anyone who can make them work, for the good of Linux. I use Rosegarden on Linux, and for various reasons, the port to Windows just doesn't cut the mustard. Any help you can provide would be greatly appreciated. :) -Scott

4 4

[alsa-devel] [RFC][PATCH 0/3 v2] firewire-lib: support Linux tracing framework to dump a part of packet data
by Takashi Sakamoto 15 Apr '16

15 Apr '16

Hi, This patchset is to update my previous one: [alsa-devel] [RFC][PATCH] ALSA: firewire-lib: support Linux tracing to dump a part of packet data http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106146.html Changes: - Drop condition statement enabled only on tracing mode. - Compute cycle count correctly and report. - Add more parameters to report. - Node IDs of source/destination. - The value of isochronous channel to use. - Add parse script written by Python3. The motivation of this patchset is to check timestamp processing of ALSA firewire stack, as writing in this message. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105608.html As a result, I confirm it compliant to IEC 61883-1/6 in these points: - Transfer 8,000 packets per second of presentation timestamp. - Transfer the same number of data blocks as sampling transfer frequency per second of presentation timestamp. - The gap of sequence of calculated presentation timestamp correctly describes the elapsed time of data blocks in a packet. - The offset between cycle start and presentation timestamp is proper, including transfer delay. Therefore, I believe that issues on dice-based models come from specific quirks, not due to bugs of ALSA firewire stack. I guess that the presentation timestamps for the models should be somewhat calculated with cycle count or the other parameters. Current implementation calculates it ignoring the cycle count. Further investigation is required to solve it. Regards Takashi Sakamoto (3): firewire-lib: compute the value of second field in cycle count for IT context firewire-lib: compute the value of second field in cycle count for IR context firewire-lib: support Linux tracing to dump a part of packet data sound/firewire/Makefile | 3 ++ sound/firewire/amdtp-stream-trace.h | 98 +++++++++++++++++++++++++++++++++++++ sound/firewire/amdtp-stream.c | 75 +++++++++++++++++++++------- 3 files changed, 159 insertions(+), 17 deletions(-) create mode 100644 sound/firewire/amdtp-stream-trace.h -- 2.7.4

1 3

Re: [alsa-devel] ALSA: lx6464es - driver for the digigram lx6464es interface
by Dan Carpenter 15 Apr '16

15 Apr '16

Hello Tim Blechmann, The patch 02bec4904508: "ALSA: lx6464es - driver for the digigram lx6464es interface" from Mar 24, 2009, leads to the following static checker warning: sound/pci/lx6464es/lx_core.c:647 lx_pipe_wait_for_state() error: uninitialized symbol'current_state'. sound/pci/lx6464es/lx_core.c 612 int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate) 613 { 614 int err; 615 u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); 616 617 mutex_lock(&chip->msg_lock); 618 lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT); 619 620 chip->rmh.cmd[0] |= pipe_cmd; 621 622 err = lx_message_send_atomic(chip, &chip->rmh); lx_message_send_atomic() returns either negative error codes or a bit mask. 623 624 if (err != 0) ^^^^^^^^ This code assumes that all non-zero values are errors which seems not true since it could be a bit mask. 625 dev_err(chip->card->dev, "could not query pipe's state\n"); 626 else 627 *rstate = (chip->rmh.stat[0] >> PSTATE_OFFSET) & 0x0F; 628 629 mutex_unlock(&chip->msg_lock); 630 return err; 631 } 632 633 static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe, 634 int is_capture, u16 state) 635 { 636 int i; 637 638 /* max 2*PCMOnlyGranularity = 2*1024 at 44100 = < 50 ms: 639 * timeout 50 ms */ 640 for (i = 0; i != 50; ++i) { 641 u16 current_state; 642 int err = lx_pipe_state(chip, pipe, is_capture, &current_state); 643 644 if (err < 0) ^^^^^^^ This code assume that only negatives are errors. 645 return err; 646 647 if (current_state == state) ^^^^^^^^^^^^^^^^^^^^^^ This is potentially slightly buggy because we returned a bit mask so we didn't initialize it. It not terribly harmful but the warning message is annoying. 648 return 0; 649 650 mdelay(1); 651 } 652 653 return -ETIMEDOUT; 654 } regards, dan carpenter

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by paul.handrigan＠cirrus.com 15 Apr '16

15 Apr '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 54 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1304 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1586 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..d18181c --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,54 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2015 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* GPIO for IRQ */ + int gpio_irq; + + /* irq line */ + int irq; + + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 907c804..f463fa5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -45,6 +45,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -372,6 +373,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712..40f73fa 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -243,6 +244,7 @@ obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..6e4dc08 --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1304 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 +#define CS35L33_INT_ATTEMPTS 20 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct regmap *cs35l33_get_regmap(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + return cs35l33->regmap; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .get_regmap = cs35l33_get_regmap, + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "release-rate", &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "hd-rm", &val32) >= 0) + hg_config->hd_rm = val32; + if (of_property_read_u32(hg, "ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "ldo-path-disable", &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "ldo-entry-delay", &val32) >= 0) + hg_config->ldo_entry_delay = val32; + hg_config->vp_hg_auto = of_property_read_bool(hg, "vp-hg-auto"); + if (of_property_read_u32(hg, "vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + int i; + + for (i = 0; i < CS35L33_INT_ATTEMPTS; i++) { + /* ack the irq by reading both status registers */ + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit at the exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + break; + + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_err(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_err(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + if (CS35L33_VBSTMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + } + + if (i == CS35L33_INT_ATTEMPTS) { + dev_err(codec->dev, + "ERROR: Interrupt handler exceeded max attpmpts\n"); + return -EINVAL; + } + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = request_threaded_irq(i2c_client->irq, NULL, cs35l33_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs35l33", cs35l33); + if (ret != 0) + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .owner = THIS_MODULE, + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..2f31952 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

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Re: [alsa-devel] [PATCH 4/4] ALSA: dice: force to add two pcm devices for listed models
by Stefan Richter 14 Apr '16

14 Apr '16

On Mar 07 Takashi Sakamoto wrote: > On Mar 7 2016 22:43, Stefan Richter wrote: > > On Mar 07 Takashi Sakamoto wrote: > >> * Dice II units > >> * TC Electronic ImpactTwin > >> * TC Electronic Konnekt24D > >> * TC Electronic Konnekt8 > >> * Dice Jr. units > >> * TC Electronic DesktopKonnect6 > >> * Dice Mini units > >> * Focusrite Saffire Pro 26 > > > > FWIW, Saffire PRO 24 and 40 are based on Dice Junior according to FFADO's > > device database. If desired I could open up my 24 and 40 to check. > > If you don't mind it, please. > (Because once opening your unit, usually, you cannot get repair service > from sellers.) > > And I wrote wrong for Dice Jr./Mini units. Correctly: > * Dice Jr. units > * TC Electronic DesktopKonnect6 > * Focusrite Saffire Pro 26 > * I have no units with Dice Mini (Oh...). > > Anyway, all of my units with Dice Jr. works fine. So we can judge that > Pro 40 has its own quirk which we don't know. Written on the Dice chip in my Saffire PRO 24, purchased in 2010: TCD2210 K34FAAB C0906 So that's a DICE Mini. And the Saffire PRO 40, purchased in early 2015: TCD2220 K8GVUF A1246 i.e. DICE JR. -- Stefan Richter -======----- -=-- -=--= http://arcgraph.de/sr/

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[alsa-devel] Tascam US-122L - Corrupt USB descriptor
by Simon Wood 14 Apr '16

14 Apr '16

Hi all, I have been struggling for the past few days to get a Tascam US-122L (USB sound-card/midi interface) working, despite reading numerous forum postings I have only been able to get the midi portion working. I note that the USB descriptor seems to be corrupt. It declares 2 interfaces, but then describes 3 separate with the same interface number for the last 2... could it be that this is confusing the Linux USB stack? I have attached a copy of the descriptor and an annotated 'lsusb -vv'. >From my experience with the HID subsystem, they have a mechanism for patching HID descriptors, is the same possible for the base USB descriptor? Trolling through 'drivers/usb' I didn't see anything... any suggestions? If this is possible I can try to increase the interface count and renumber the last one. Cheers, Simon. PS. For reference the device is: Model No: US-122L Serial: (21)0120xxx Other barcode: 043774021628

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[alsa-devel] [PATCH] ALSA: pcm : Call kill_fasync() in stream lock
by Takashi Iwai 14 Apr '16

14 Apr '16

Currently kill_fasync() is called outside the stream lock in snd_pcm_period_elapsed(). This is potentially racy, since the stream may get released even during the irq handler is running. Although snd_pcm_release_substream() calls snd_pcm_drop(), this doesn't guarantee that the irq handler finishes, thus the kill_fasync() call outside the stream spin lock may be invoked after the substream is detached, as recently reported by KASAN. As a quick workaround, move kill_fasync() call inside the stream lock. The fasync is rarely used interface, so this shouldn't have a big impact from the performance POV. Ideally, we should implement some sync mechanism for the proper finish of stream and irq handler. But this oneliner should suffice for most cases, so far. Reported-by: Baozeng Ding <sploving1(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 3a9b66c6e09c..0aca39762ed0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_timer_interrupt(substream->timer, 1); #endif _end: - snd_pcm_stream_unlock_irqrestore(substream, flags); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_pcm_stream_unlock_irqrestore(substream, flags); } EXPORT_SYMBOL(snd_pcm_period_elapsed); -- 2.8.1

1 0

[alsa-devel] EBADFD caused by commit dec428c352217010e4b8bd750d302b8062339d32
by Lars Lindqvist 14 Apr '16

14 Apr '16

Hi! Since alsa-lib commit dec428c352217010e4b8bd750d302b8062339d32, I've occationally been hit by an EBADFD whenever any program tries to play sound. The situation I get is that the first shmget succeds, so dmix->shmid >= 0, therefore first_instance = 0. But buf.shm_nattach = 1, so before the commit shmptr would have been zeroed out, but isn't anymore. And since I still have: dmix->shmptr->magic == SND_PCM_DIRECT_MAGIC, I don't get EINVAL, but EBADFD, somewhere down the line. >From what I understand, the race condition that was fixed would still be avoided if shmptr was zeroed on (first_instance || buf.shm_nattch == 1). If that is the case, would you please consider applying attached diff? Regards, Lars Lindqvist

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[alsa-devel] [PATCH 0/1] ASoC: sti: select player for I2S/TDM TX bus
by Moise Gergaud 14 Apr '16

14 Apr '16

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/106391.html Moise Gergaud (1): ASoC: sti: select player for I2S/TDM TX bus sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 42 +++++++++++++++++++++++++++-------------- 2 files changed, 29 insertions(+), 14 deletions(-) -- 1.9.1

2 2

[alsa-devel] Multi-channel playback
by Li, Steve 14 Apr '16

14 Apr '16

Hi, I am trying to playback a multi-channel PCM stream, e.g. 5.1. Is there an API for configuring the channel positions? I found this API: int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); typedef struct snd_pcm_chmap { unsigned int channels; unsigned int pos[0]; } snd_pcm_chmap_t; But when I call it, it returns -6. Also, calling snd_pcm_get_chmap() return NULL. Your help is appreciated. Thanks Steve ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

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[alsa-devel] [RFC PATCH 0/4] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 14 Apr '16

14 Apr '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. The interface to connect with BT chip is not finish yet, but as an upstream rookie I really need some comments. Any comments will be greatly appreciated. Garlic Tseng (4): ASoC: mediatek: Refine mt8173 driver ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: mediatek: Add mt2701-cs42448 driver and config option. .../devicetree/bindings/sound/mt2701-cs42448.txt | 39 + .../devicetree/bindings/sound/mtk2701-afe-pcm.txt | 150 ++ sound/soc/mediatek/Kconfig | 21 + sound/soc/mediatek/Makefile | 15 +- sound/soc/mediatek/mt2701/Makefile | 20 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 376 +++++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c | 259 +++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h | 28 + sound/soc/mediatek/mt2701/mtk2701-afe-common.h | 217 +++ sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c | 1752 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mtk2701-irq.c | 109 ++ sound/soc/mediatek/mt2701/mtk2701-irq.h | 30 + sound/soc/mediatek/mt2701/mtk2701-reg.h | 165 ++ sound/soc/mediatek/mt8173-max98090.c | 213 --- sound/soc/mediatek/mt8173-rt5650-rt5514.c | 258 --- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 288 ---- sound/soc/mediatek/mt8173-rt5650.c | 236 --- sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-max98090.c | 213 +++ sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 258 +++ sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 288 ++++ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 236 +++ sound/soc/mediatek/mt8173/mtk-afe-common.h | 101 ++ sound/soc/mediatek/mt8173/mtk-afe-pcm.c | 1333 +++++++++++++++ sound/soc/mediatek/mtk-afe-common.h | 101 -- sound/soc/mediatek/mtk-afe-pcm.c | 1333 --------------- 26 files changed, 5610 insertions(+), 2436 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 Documentation/devicetree/bindings/sound/mtk2701-afe-pcm.txt create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-reg.h delete mode 100644 sound/soc/mediatek/mt8173-max98090.c delete mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5514.c delete mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5676.c delete mode 100644 sound/soc/mediatek/mt8173-rt5650.c create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-max98090.c create mode 100644 sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c create mode 100644 sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c create mode 100644 sound/soc/mediatek/mt8173/mt8173-rt5650.c create mode 100644 sound/soc/mediatek/mt8173/mtk-afe-common.h create mode 100644 sound/soc/mediatek/mt8173/mtk-afe-pcm.c delete mode 100644 sound/soc/mediatek/mtk-afe-common.h delete mode 100644 sound/soc/mediatek/mtk-afe-pcm.c -- 1.9.1

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[alsa-devel] [PATCH] of: add missing const for of_parse_phandle_with_args() in !CONFIG_OF
by Kuninori Morimoto 14 Apr '16

14 Apr '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 93c667ca2598bd84f1bd3f2fa176af69707699fe ("of: *node argument to of_parse_phandle_with_args should be const") changed struct device node *np as "const". but it cares CONFIG_OF case only, !CONFIG_OF case need it too. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/linux/of.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/of.h b/include/linux/of.h index 7fcb681..4f2ce1b 100644 --- a/include/linux/of.h +++ b/include/linux/of.h @@ -585,7 +585,7 @@ static inline struct device_node *of_parse_phandle(const struct device_node *np, return NULL; } -static inline int of_parse_phandle_with_args(struct device_node *np, +static inline int of_parse_phandle_with_args(const struct device_node *np, const char *list_name, const char *cells_name, int index, -- 1.9.1

1 0

[alsa-devel] [PATCH 0/2 v5] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 13 Apr '16

13 Apr '16

Hi all, This is v5 of these patches. In this version I dropped the ADV7511 audio patch because, quoting Lars-Peter Clausen: "The reason why this driver is still out of tree, is because there has been no conclusion yet on how to go forward with the whole HDMI integration. So this is not going to get merged until that issue has been addressed." So, until that issue is addressed I will not send any ADV7511 audio related patches but if for some reason anyone wants to merge it I can send the pathes again. V4, which includes the ADV7511 audio patches is available at: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106650.html Looking forward to your comments! Best regards, Jose Miguel Abreu ----- ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds I2S audio for the AXS10x platform. Changes v4 -> v5 * Resolve undefined references when compiling as module * Dropped adv7511 audio patches * Use DMA properties in I2S to check which mode to use: PIO or DMA (as suggested by Lars-Peter Clausen) Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Jose Abreu (2): ASoC: dwc: Add custom PCM driver ASoC: dwc: Update DOCUMENTATION for I2S Driver .../devicetree/bindings/sound/designware-i2s.txt | 9 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware.h | 71 +++++++ sound/soc/dwc/designware_i2s.c | 94 ++++++--- sound/soc/dwc/designware_pcm.c | 230 +++++++++++++++++++++ 6 files changed, 385 insertions(+), 29 deletions(-) create mode 100644 sound/soc/dwc/designware.h create mode 100644 sound/soc/dwc/designware_pcm.c -- 1.9.1

2 3

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix regression of monitor_present flag in eld proc file
by Takashi Iwai 13 Apr '16

13 Apr '16

From: Hyungwon Hwang <hyungwon.hwang7(a)gmail.com> The commit [bd48128539ab: ALSA: hda - Fix forgotten HDMI monitor_present update] covered the missing update of monitor_present flag, but this caused a regression for devices without the i915 eld notifier. Since the old code supposed that pin_eld->monitor_present was updated by the caller side, the hdmi_present_sense_via_verbs() doesn't update the temporary eld->monitor_present but only pin_eld->monitor_present, which is now overridden in update_eld(). The fix is to update pin_eld->monitor_present as well before calling update_eld(). Note that this may still leave monitor_present flag in an inconsistent state when the driver repolls, but this is at least the old behavior. More proper fix will follow in the later patch. Fixes: bd48128539ab ('ALSA: hda - Fix forgotten HDMI monitor_present update') Signed-off-by: Hyungwon Hwang <hyungwon.hwang7(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 5af372d01834..9de114df270b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1414,6 +1414,8 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, mutex_lock(&per_pin->lock); pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + eld->monitor_present = pin_eld->monitor_present; + if (pin_eld->monitor_present) eld->eld_valid = !!(present & AC_PINSENSE_ELDV); else -- 2.8.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Update HDMI monitor_present from a temporary structure
by Hyungwon Hwang 13 Apr '16

13 Apr '16

pin_eld is a temporary structure. So before calling update_eld(), eld should be updated from pin_eld. Signed-off-by: Hyungwon Hwang <hyungwon.hwang7(a)gmail.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 5af372d..9de114d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1414,6 +1414,8 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, mutex_lock(&per_pin->lock); pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + eld->monitor_present = pin_eld->monitor_present; + if (pin_eld->monitor_present) eld->eld_valid = !!(present & AC_PINSENSE_ELDV); else -- 2.5.5

2 3

[alsa-devel] [PATCH v5] ucm: add cset-tlv
by Hsin-Yu Chao 13 Apr '16

13 Apr '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. The TLV file to set to kcontrol will be checked first by file size not larger than 16 MB, and then examine if the length field reports correct number of bytes in the TLV file. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 94 +++++++++++++++++++++++++++++++++++++++++++++-------- src/ucm/parser.c | 10 ++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 92 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..24d9510 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,57 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int read_tlv_file(unsigned int **res, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + struct snd_ctl_tlv *tlv; + + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (fstat(fd, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + if (sz > 16 * 1024 * 1024 || sz < 8 || sz % 4) { + uc_error("File size should be less than 16 MB " + "and multiple of 4"); + err = -EINVAL; + goto __fail; + } + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -EIO; + free(*res); + *res = NULL; + } + /* Check if the tlv file specifies valid size. */ + tlv = (struct snd_ctl_tlv *)(*res); + if (tlv->length + 2 * sizeof(unsigned int) != sz) { + uc_error("Invalid tlv size: %d", tlv->length); + err = -EINVAL; + free(*res); + *res = NULL; + } + +__fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +277,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + unsigned int *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +293,36 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + goto __fail; + } + err = read_tlv_file(&res, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +331,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +365,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.1.2

3 2

[alsa-devel] [PATCH v4] ucm: add cset-tlv
by Hsin-Yu Chao 13 Apr '16

13 Apr '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. The TLV file to set to kcontrol will be checked first by file size not larger than 16 MB, and then examine if the length field reports correct number of bytes in the TLV file. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 93 +++++++++++++++++++++++++++++++++++++++++++++-------- src/ucm/parser.c | 10 ++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 91 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..cf9e825 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,56 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int read_tlv_file(char **res, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + unsigned int *tlv; + + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (fstat(fd, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + if (sz > 16 * 1024 * 1024 || sz < 8) { + uc_error("File size should be less than 16 MB"); + err = -EINVAL; + goto __fail; + } + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -EIO; + free(*res); + *res = NULL; + } + /* Check if the tlv file specifies valid size. */ + tlv = (unsigned int *)(*res); + if (tlv[1] + 2 * sizeof(unsigned int) != sz) { + uc_error("Invalid tlv size"); + err = -EINVAL; + free(*res); + *res = NULL; + } + + __fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +276,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + char *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +292,36 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + goto __fail; + } + err = read_tlv_file(&res, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, (unsigned int *)res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +330,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +364,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.1.2

3 4

[alsa-devel] [PATCH v3] ucm: add cset-tlv
by Hsin-Yu Chao 13 Apr '16

13 Apr '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. The TLV file to set to kcontrol will be checked first by file size not larger than 16 MB, and then examine if the length field reports correct number of bytes in the TLV file. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 91 +++++++++++++++++++++++++++++++++++++++++++++-------- src/ucm/parser.c | 10 ++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 89 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..d0929a2 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,54 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int read_tlv_file(char **res, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + unsigned int *tlv; + + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + if (sz > 16 * 1024 * 1024 || sz < 8) { + uc_error("File size should be less than 16 MB"); + goto __fail; + } + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + free(*res); + *res = NULL; + } + /* Check if the tlv file specifies valid size. */ + tlv = (unsigned int *)(*res); + if (tlv[1] + 2 * sizeof(unsigned int) != sz) { + uc_error("Invalid tlv size"); + free(*res); + *res = NULL; + } + + __fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +274,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + char *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +290,36 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + goto __fail; + } + err = read_tlv_file(&res, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, (unsigned int *)res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +328,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +362,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.8.0.rc3.226.g39d4020

2 2

[alsa-devel] [PATCH] ALSA: hda - Remove temporary buffer for ELD update
by Hyungwon Hwang 13 Apr '16

13 Apr '16

With the introduce of pcm_lock and per_pin->lock, this temporary buffer is not needed anymore. Signed-off-by: Hyungwon Hwang <hyungwon.hwang7(a)gmail.com> --- sound/pci/hda/patch_hdmi.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..a24d5b4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1395,8 +1395,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, struct hda_jack_tbl *jack; struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld = &spec->temp_eld; - struct hdmi_eld *pin_eld = &per_pin->sink_eld; + struct hdmi_eld *eld = &per_pin->sink_eld; hda_nid_t pin_nid = per_pin->pin_nid; /* * Always execute a GetPinSense verb here, even when called from @@ -1413,15 +1412,15 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, present = snd_hda_pin_sense(codec, pin_nid); mutex_lock(&per_pin->lock); - pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); - if (pin_eld->monitor_present) + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + if (eld->monitor_present) eld->eld_valid = !!(present & AC_PINSENSE_ELDV); else eld->eld_valid = false; codec_dbg(codec, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (spec->ops.pin_get_eld(codec, pin_nid, eld->eld_buffer, @@ -1441,7 +1440,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, else update_eld(codec, per_pin, eld); - ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid; + ret = !repoll || !eld->monitor_present || eld->eld_valid; jack = snd_hda_jack_tbl_get(codec, pin_nid); if (jack) -- 2.5.0

2 2

[alsa-devel] [PATCH 1/1] ASoC: au1x: use correct format specifier
by Heinrich Schuchardt 13 Apr '16

13 Apr '16

Documentation/printk-formats.txt has unsigned long: use %lu or %lx size_t: use %zu or %zx runtime->dma_bytes is of type size_t. runtime->min_align is of type unsigned long. Signed-off-by: Heinrich Schuchardt <xypron.glpk(a)gmx.de> --- sound/soc/au1x/dbdma2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 5741c0a..b5d1caa 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, stype = substream->stream; pcd = to_dmadata(substream); - DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " - "runtime->min_align %d\n", + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu " + "runtime->min_align %lu\n", (unsigned long)runtime->dma_area, (unsigned long)runtime->dma_addr, runtime->dma_bytes, runtime->min_align); -- 2.1.4

1 0

[alsa-devel] [PATCH 1/1] ASoC: fsl: imx-pcm-fiq: use correct format specifier
by Heinrich Schuchardt 13 Apr '16

13 Apr '16

Documentation/printk-formats.txt has size_t: use %zu or %zx runtime->dma_bytes is of type size_t. Signed-off-by: Heinrich Schuchardt <xypron.glpk(a)gmx.de> --- sound/soc/fsl/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index e63cd5e..dac6688 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret, + pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret, runtime->dma_area, &runtime->dma_addr, runtime->dma_bytes); -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Update documentation
by Takashi Iwai 12 Apr '16

12 Apr '16

Update the URLs for alsa-info.sh and hda-emu. Also drop the alsa-driver snapshot URL since it's been discontinued recently. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio.txt | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index e7193aac669c..d4510ebf2e8c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -655,17 +655,6 @@ development branches in general while the development for the current and next kernels are found in for-linus and for-next branches, respectively. -If you are using the latest Linus tree, it'd be better to pull the -above GIT tree onto it. If you are using the older kernels, an easy -way to try the latest ALSA code is to build from the snapshot -tarball. There are daily tarballs and the latest snapshot tarball. -All can be built just like normal alsa-driver release packages, that -is, installed via the usual spells: configure, make and make -install(-modules). See INSTALL in the package. The snapshot tarballs -are found at: - -- ftp://ftp.suse.com/pub/people/tiwai/snapshot/ - Sending a Bug Report ~~~~~~~~~~~~~~~~~~~~ @@ -699,7 +688,12 @@ problems. alsa-info ~~~~~~~~~ The script `alsa-info.sh` is a very useful tool to gather the audio -device information. You can fetch the latest version from: +device information. It's included in alsa-utils package. The latest +version can be found on git repository: + +- git://git.alsa-project.org/alsa-utils.git + +The script can be fetched directly from the following URL, too: - http://www.alsa-project.org/alsa-info.sh @@ -836,15 +830,11 @@ can get a proc-file dump at the current state, get a list of control (mixer) elements, set/get the control element value, simulate the PCM operation, the jack plugging simulation, etc. -The package is found in: - -- ftp://ftp.suse.com/pub/people/tiwai/misc/ - -A git repository is available: +The program is found in the git repository below: - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git -See README file in the tarball for more details about hda-emu +See README file in the repository for more details about hda-emu program. -- 2.8.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Add missing capture_hook calls for dyn-ADC PCM streams
by Takashi Iwai 12 Apr '16

12 Apr '16

The calls for capture_hook were missing in dyn_adc_capture_pcm_prepare and cleanup callbacks. Luckily there are no users of the capture hooks with dyn-adc PCM, so far, thus this doesn't change the behavior of existing devices, but it's a fix for a future usage. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ca5b89f088a..dc2c13687118 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5432,6 +5432,7 @@ static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_PREPARE); return 0; } @@ -5442,6 +5443,7 @@ static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_gen_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = 0; + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_CLEANUP); return 0; } -- 2.8.1

1 0

[alsa-devel] [PATCH v2 0/7] ASoC: sti: unip tdm mode
by Moise Gergaud 12 Apr '16

12 Apr '16

This patchset is V2 of "ASoC: sti: tdm: uni tdm mode" V2 consists of splitting the V1 patch into 7 patches. Moise Gergaud (7): ASoC: sti: macro for uniperif tdm regs access ASoC: sti: rename unip player type into common player & reader type ASoC: sti: define tdm type & default tdm hw config ASoC: sti: helper functions for unip tdm slots configuration ASoC: sti: helper functions to fix tdm runtime params ASoC: sti: unip player tdm mode ASoC: sti: unip reader tdm mode sound/soc/sti/sti_uniperif.c | 144 ++++++++++++++++++++++++- sound/soc/sti/uniperif.h | 197 +++++++++++++++++++++++++++++++++- sound/soc/sti/uniperif_player.c | 140 +++++++++++++++++------- sound/soc/sti/uniperif_reader.c | 229 +++++++++++++++++++++++++++++----------- 4 files changed, 606 insertions(+), 104 deletions(-) -- 1.9.1

2 14

[alsa-devel] [PATCH v2] ASoC: rockchip: i2s: configure the sdio pins' iomux mode
by Sugar Zhang 12 Apr '16

12 Apr '16

There are 3 i2s sdio pins, which iomux mode is as follows: - sdi3_sdo1 - sdi2_sdo2 - sdi1_sdo3 we need to configure these pins' iomux mode via the GRF register when use multi channel playback/capture. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - fix a typo error. as suggested by Heiko: - use the general grf-binding without any offset - keep the actual offset in the driver on a per-soc basis. .../devicetree/bindings/sound/rockchip-i2s.txt | 5 ++ sound/soc/rockchip/rockchip_i2s.c | 67 +++++++++++++++++++--- sound/soc/rockchip/rockchip_i2s.h | 7 +++ 3 files changed, 70 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 6e86d8a..4ea29aa 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -23,6 +23,11 @@ Required properties: - rockchip,playback-channels: max playback channels, if not set, 8 channels default. - rockchip,capture-channels: max capture channels, if not set, 2 channels default. +Required properties for controller which support multi channels +playback/capture: + +- rockchip,grf: the phandle of the syscon node for GRF register. + Example for rk3288 I2S controller: i2s@ff890000 { diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e204..d4f7eac 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -11,8 +11,10 @@ */ #include <linux/module.h> +#include <linux/mfd/syscon.h> #include <linux/delay.h> #include <linux/of_gpio.h> +#include <linux/of_device.h> #include <linux/clk.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> @@ -23,6 +25,11 @@ #define DRV_NAME "rockchip-i2s" +struct rk_i2s_pins { + u32 reg_offset; + u32 shift; +}; + struct rk_i2s_dev { struct device *dev; @@ -33,8 +40,10 @@ struct rk_i2s_dev { struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; + struct regmap *grf; bool is_master_mode; + const struct rk_i2s_pins *pins; }; static int i2s_runtime_suspend(struct device *dev) @@ -277,6 +286,30 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK, val); + if (!IS_ERR(i2s->grf) && i2s->pins) { + regmap_read(i2s->regmap, I2S_TXCR, &val); + val &= I2S_TXCR_CSR_MASK; + + switch (val) { + case I2S_CHN_4: + val = I2S_IO_4CH_OUT_6CH_IN; + break; + case I2S_CHN_6: + val = I2S_IO_6CH_OUT_4CH_IN; + break; + case I2S_CHN_8: + val = I2S_IO_8CH_OUT_2CH_IN; + break; + default: + val = I2S_IO_2CH_OUT_8CH_IN; + break; + } + + val <<= i2s->pins->shift; + val |= (I2S_IO_DIRECTION_MASK << i2s->pins->shift) << 16; + regmap_write(i2s->grf, i2s->pins->reg_offset, val); + } + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, I2S_DMACR_TDL(16)); regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, @@ -462,9 +495,23 @@ static const struct regmap_config rockchip_i2s_regmap_config = { .cache_type = REGCACHE_FLAT, }; +static const struct rk_i2s_pins rk3399_i2s_pins = { + .reg_offset = 0xe220, + .shift = 11, +}; + +static const struct of_device_id rockchip_i2s_match[] = { + { .compatible = "rockchip,rk3066-i2s", }, + { .compatible = "rockchip,rk3188-i2s", }, + { .compatible = "rockchip,rk3288-i2s", }, + { .compatible = "rockchip,rk3399-i2s", .data = &rk3399_i2s_pins }, + {}, +}; + static int rockchip_i2s_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; + const struct of_device_id *of_id; struct rk_i2s_dev *i2s; struct snd_soc_dai_driver *soc_dai; struct resource *res; @@ -478,6 +525,17 @@ static int rockchip_i2s_probe(struct platform_device *pdev) return -ENOMEM; } + i2s->dev = &pdev->dev; + + i2s->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); + if (!IS_ERR(i2s->grf)) { + of_id = of_match_device(rockchip_i2s_match, &pdev->dev); + if (!of_id || !of_id->data) + return -EINVAL; + + i2s->pins = of_id->data; + } + /* try to prepare related clocks */ i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk"); if (IS_ERR(i2s->hclk)) { @@ -517,7 +575,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; i2s->capture_dma_data.maxburst = 4; - i2s->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, i2s); pm_runtime_enable(&pdev->dev); @@ -583,14 +640,6 @@ static int rockchip_i2s_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id rockchip_i2s_match[] = { - { .compatible = "rockchip,rk3066-i2s", }, - { .compatible = "rockchip,rk3188-i2s", }, - { .compatible = "rockchip,rk3288-i2s", }, - { .compatible = "rockchip,rk3399-i2s", }, - {}, -}; - static const struct dev_pm_ops rockchip_i2s_pm_ops = { SET_RUNTIME_PM_OPS(i2s_runtime_suspend, i2s_runtime_resume, NULL) diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index dc6e2c7..8e239d3 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -236,4 +236,11 @@ enum { #define I2S_TXDR (0x0024) #define I2S_RXDR (0x0028) +/* io direction cfg register */ +#define I2S_IO_DIRECTION_MASK (7) +#define I2S_IO_8CH_OUT_2CH_IN (0) +#define I2S_IO_6CH_OUT_4CH_IN (4) +#define I2S_IO_4CH_OUT_6CH_IN (6) +#define I2S_IO_2CH_OUT_8CH_IN (7) + #endif /* _ROCKCHIP_IIS_H */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Add DSP firmware manifest parsing
by Vinod Koul 12 Apr '16

12 Apr '16

From: "Nc,Shreyas" <shreyas.nc(a)intel.com> Module params like module_id and loadable flag can changed in the DSP Firmware. These are kept in the firmware manifest and driver should read these values from this manifest. So, add support to parse the DSP firmware manifest and read these module params. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 134 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 109 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 4 + sound/soc/intel/skylake/skl-sst.c | 11 ++- sound/soc/intel/skylake/skl-topology.c | 15 ++-- 6 files changed, 268 insertions(+), 7 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index c28f5d0e1d99..e2fb5b763b21 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o + skl-sst.o skl-dsp-parse.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-dsp-parse.c b/sound/soc/intel/skylake/skl-dsp-parse.c new file mode 100644 index 000000000000..1736dec68e3c --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.c @@ -0,0 +1,134 @@ +/* + * skl-dsp-parse.c - Implements DSP firmware manifest parsing + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <asm/types.h> +#include <linux/types.h> +#include <linux/string.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/slab.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-tplg-interface.h" +#include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" + +/* + * Get the module id for the module by checking + * the table for the UUID for the module + */ +int snd_skl_get_module_info(struct skl_sst *ctx, char *uuid, + struct skl_dfw_module *dfw_config) +{ + int i, num; + struct uuid_tbl *tbl; + + tbl = ctx->tbl; + num = ctx->num_modules; + + for (i = 0; i < num; i++) { + if (!strcmp(uuid, tbl[i].uuid)) { + dfw_config->module_id = tbl[i].module_id; + dfw_config->is_loadable = tbl[i].is_loadable; + + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_skl_get_module_info); + +/* + * Get the uuid from the DSP firmware structure in a string + * of the format XXXXXXXX-XXXX-XXXX-XXXXXXXXXXXX + */ +static char *get_canonical_uuid(char *buf, + struct adsp_module_entry *entry) +{ + unsigned int i; + + sprintf(buf, "%08X", entry->uuid.id_1); + sprintf(buf+strlen(buf), "-"); + + for (i = 0; i < ARRAY_SIZE(entry->uuid.id_2); i++) { + sprintf(buf+strlen(buf), + "%04X", entry->uuid.id_2[i]); + sprintf(buf+strlen(buf), "-"); + } + + for (i = 0; i < ARRAY_SIZE(entry->uuid.id_3); i++) { + sprintf(buf+strlen(buf), + "%02X", entry->uuid.id_3[i]); + if (i == 1) + sprintf(buf+strlen(buf), "-"); + } + + return buf; +} + +/* + * Parse the firmware binary to get the UUID, module id + * and loadable flags + */ +int parse_fw_bin(struct sst_dsp *ctx) +{ + struct adsp_fw_hdr *adsp_hdr; + struct adsp_module_entry *mod_entry; + int i, num_entry; + char uuid_str[UUID_STR_SIZE]; + const char *buf; + struct skl_sst *skl = ctx->thread_context; + struct uuid_tbl *tbl; + + + /* Get the FW pointer to derive ADSP header */ + buf = ctx->fw->data; + + adsp_hdr = (struct adsp_fw_hdr *)(buf + SKL_ADSP_FW_BIN_HDR_OFFSET); + + mod_entry = (struct adsp_module_entry *) + (buf + SKL_ADSP_FW_BIN_HDR_OFFSET + adsp_hdr->header_len); + + num_entry = adsp_hdr->num_module_entries; + + tbl = devm_kzalloc(ctx->dev, + num_entry * sizeof(struct uuid_tbl), GFP_KERNEL); + + if (!tbl) + return -ENOMEM; + + /* + * Populate the UUID table to store module_id + * and loadable flags for the module + */ + + for (i = 0; i < num_entry; i++, mod_entry++) { + get_canonical_uuid(uuid_str, mod_entry); + + strncpy(tbl[i].uuid, uuid_str, strlen(uuid_str)); + + tbl[i].module_id = i; + tbl[i].is_loadable = mod_entry->type.load_type; + } + + skl->tbl = tbl; + skl->num_modules = num_entry; + return 0; +} +EXPORT_SYMBOL_GPL(parse_fw_bin); diff --git a/sound/soc/intel/skylake/skl-dsp-parse.h b/sound/soc/intel/skylake/skl-dsp-parse.h new file mode 100644 index 000000000000..2da1bbad9145 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.h @@ -0,0 +1,109 @@ +/* + * skl-dsp-parse.h + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <asm/types.h> +#include <linux/types.h> +#include <linux/uuid.h> +#include "skl-tplg-interface.h" +#include "../common/sst-dsp-priv.h" + +#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 +#define UUID_STR_SIZE 37 +#define DEFAULT_HASH_SHA256_LEN 32 + +struct skl_dfw_module_mod { + char name[100]; + struct skl_dfw_module skl_dfw_mod; +}; + +struct UUID { + u32 id_1; + u16 id_2[2]; + u8 id_3[8]; +}; + +union seg_flags { + u32 ul; + struct { + u32 contents : 1; + u32 alloc : 1; + u32 load : 1; + u32 read_only : 1; + u32 code : 1; + u32 data : 1; + u32 _rsvd0 : 2; + u32 type : 4; + u32 _rsvd1 : 4; + u32 length : 16; + } r; +} __packed; + +struct segment_desc { + union seg_flags flags; + u32 v_base_addr; + u32 file_offset; +}; + +struct module_type { + u32 load_type : 4; + u32 auto_start : 1; + u32 domain_ll : 1; + u32 domain_dp : 1; + u32 rsvd_ : 25; +} __packed; + +struct adsp_module_entry { + u32 struct_id; + u8 name[8]; + struct UUID uuid; + struct module_type type; + u8 hash1[DEFAULT_HASH_SHA256_LEN]; + u32 entry_point; + u16 cfg_offset; + u16 cfg_count; + u32 affinity_mask; + u16 instance_max_count; + u16 instance_bss_size; + struct segment_desc segments[3]; +} __packed; + +struct adsp_fw_hdr { + u32 header_id; + u32 header_len; + u8 name[8]; + u32 preload_page_count; + u32 fw_image_flags; + u32 feature_mask; + u16 major_version; + u16 minor_version; + u16 hotfix_version; + u16 build_version; + u32 num_module_entries; + u32 hw_buf_base_addr; + u32 hw_buf_length; + u32 load_offset; +} __packed; + +struct uuid_tbl { + char uuid[UUID_STR_SIZE]; + int module_id; + int is_loadable; +}; + +int parse_fw_bin(struct sst_dsp *ctx); +int snd_skl_get_module_info(struct skl_sst *ctx, char *uuid, + struct skl_dfw_module *dfw_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index d59d1ba62a43..67d56be8cb77 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -60,6 +60,10 @@ struct skl_sst { void (*enable_miscbdcge)(struct device *dev, bool enable); /*Is CGCTL.MISCBDCGE disabled*/ bool miscbdcg_disabled; + + /* Populate module information */ + struct uuid_tbl *tbl; + int num_modules; }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 348a734f8e24..9fbe922d05ce 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -24,6 +24,7 @@ #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" #include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" #define SKL_BASEFW_TIMEOUT 300 #define SKL_INIT_TIMEOUT 1000 @@ -69,7 +70,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { - int ret = 0, i; + int ret = 0, i, err; struct skl_sst *skl = ctx->thread_context; u32 reg; @@ -83,6 +84,14 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) skl_dsp_disable_core(ctx); return -EIO; } + + err = parse_fw_bin(ctx); + if (err < 0) { + dev_err(ctx->dev, + "Firmware parsing err: %d\n", ret); + release_firmware(ctx->fw); + return err; + } } ret = skl_dsp_boot(ctx); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 545b4e77b8aa..d64edfa45b6e 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -28,6 +28,7 @@ #include "skl-tplg-interface.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "skl-dsp-parse.h" #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) @@ -1550,7 +1551,15 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, return -ENOMEM; w->priv = mconfig; + memcpy(mconfig->guid, dfw_config->uuid, + ARRAY_SIZE(dfw_config->uuid)); + + ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config); + if (ret < 0) + return ret; + mconfig->id.module_id = dfw_config->module_id; + mconfig->is_loadable = dfw_config->is_loadable; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; mconfig->ibs = dfw_config->ibs; @@ -1558,7 +1567,7 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, mconfig->core_id = dfw_config->core_id; mconfig->max_in_queue = dfw_config->max_in_queue; mconfig->max_out_queue = dfw_config->max_out_queue; - mconfig->is_loadable = dfw_config->is_loadable; + skl_tplg_fill_fmt(mconfig->in_fmt, dfw_config->in_fmt, MODULE_MAX_IN_PINS); skl_tplg_fill_fmt(mconfig->out_fmt, dfw_config->out_fmt, @@ -1579,10 +1588,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, mconfig->time_slot = dfw_config->time_slot; mconfig->formats_config.caps_size = dfw_config->caps.caps_size; - if (dfw_config->is_loadable) - memcpy(mconfig->guid, dfw_config->uuid, - ARRAY_SIZE(dfw_config->uuid)); - mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) * sizeof(*mconfig->m_in_pin), GFP_KERNEL); -- 2.4.3

2 2

[alsa-devel] Too fast playback everywhere, except audacity
by Giedrius Trainavičius 12 Apr '16

12 Apr '16

Hello, I am developing a driver for a DAC on raspberry pi 3 by mostly making use of existing PCM5102A, and bcm2708 I2S driver code. What I've done is just configuring the cpu dai link, and for the sampling rate changes, I am setting some gpios which configures the clock of an ADC (PCM1804) which is a master. The main CPU and the DAC are slaves to the ADC and I have a 24.576MHz oscillator. The sampling rates I want to support are 48000, 96000 and 192000. I am able to react just fine to the requested rate in hw_params callback. Trying out the different rates in audacity works just fine. However, if I try to playback any wav files using mplayer or aplay or youtube, all of their playback requests for 48000, but they get played way too fast. Also if I request mplayer to use alsa output, it gets a buffer underrun and gets stuck. If I set audacity to use some other output device than alsa directly, it plays corrupted audio and hangs on stopping playback. How does audacity manage to play the audio at the correct speed? Is it doing software resampling to the 'project rate'? I've tried getting resampling to work using 'plug', but completely unsuccessfully - I can't get any sound out of it, it says it failed setting the requested parameters. Any pointers to what I should be investigating? I am new to the linux kernel module development, as well as sound drivers. Thank you in advance, Giedrius.

1 0

[alsa-devel] [PATCH 0/5 v4] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 11 Apr '16

11 Apr '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Jose Abreu (5): drm/i2c/adv7511: Rename and move to separate folder drm/i2c/adv7511: Add audio support ASoC: dwc: Use fifo depth to program FCR ASoC: dwc: Add custom PCM driver ASoC: dwc: Update DOCUMENTATION for I2S Driver .../bindings/display/bridge/adi,adv7511.txt | 3 + .../devicetree/bindings/sound/designware-i2s.txt | 5 + drivers/gpu/drm/i2c/Kconfig | 6 +- drivers/gpu/drm/i2c/Makefile | 2 +- drivers/gpu/drm/i2c/adv7511/Kconfig | 18 ++ drivers/gpu/drm/i2c/adv7511/Makefile | 3 + drivers/gpu/drm/i2c/{ => adv7511}/adv7511.h | 53 ++++ drivers/gpu/drm/i2c/adv7511/adv7511_audio.c | 310 +++++++++++++++++++++ .../drm/i2c/{adv7511.c => adv7511/adv7511_core.c} | 43 +-- include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware.h | 70 +++++ sound/soc/dwc/designware_i2s.c | 106 +++++-- sound/soc/dwc/designware_pcm.c | 230 +++++++++++++++ 15 files changed, 796 insertions(+), 64 deletions(-) create mode 100644 drivers/gpu/drm/i2c/adv7511/Kconfig create mode 100644 drivers/gpu/drm/i2c/adv7511/Makefile rename drivers/gpu/drm/i2c/{ => adv7511}/adv7511.h (90%) create mode 100644 drivers/gpu/drm/i2c/adv7511/adv7511_audio.c rename drivers/gpu/drm/i2c/{adv7511.c => adv7511/adv7511_core.c} (97%) create mode 100644 sound/soc/dwc/designware.h create mode 100644 sound/soc/dwc/designware_pcm.c -- 1.9.1

3 21

[alsa-devel] [PATCH] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
by Sven Eckelmann 11 Apr '16

11 Apr '16

The Lenovo Thinkpad T460s requires the alc_fixup_tpt440_dock as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T460s (20F9CT01WW) using a ThinkPad Ultradock on kernel 4.4.6. Signed-off-by: Sven Eckelmann <sven(a)narfation.org> Tested-by: Simon Wunderlich <sw(a)simonwunderlich.de> --- v3: - Removed ALC292_FIXUP_TPT460_DOCK v2: - rebased on current master sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fefe83f..1402ba9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4760,6 +4760,7 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, + ALC292_FIXUP_TPT460, }; static const struct hda_fixup alc269_fixups[] = { @@ -5409,6 +5410,12 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC292_FIXUP_TPT460] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440_dock, + .chained = true, + .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5563,7 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -5658,6 +5665,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, + {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {} }; #define ALC225_STANDARD_PINS \

2 1

[alsa-devel] [PATCH v2] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
by Sven Eckelmann 11 Apr '16

11 Apr '16

The Lenovo Thinkpad T460s requires the alc_fixup_tpt440_dock as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T460s (20F9CT01WW) using a ThinkPad Ultradock on kernel 4.4.6. Signed-off-by: Sven Eckelmann <sven(a)narfation.org> Tested-by: Simon Wunderlich <sw(a)simonwunderlich.de> --- v2: - rebased on current master sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fefe83f..36d371c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4760,6 +4760,8 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, + ALC292_FIXUP_TPT460_DOCK, + ALC292_FIXUP_TPT460, }; static const struct hda_fixup alc269_fixups[] = { @@ -5409,6 +5411,18 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC292_FIXUP_TPT460_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440_dock, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, + [ALC292_FIXUP_TPT460] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC292_FIXUP_TPT460_DOCK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5563,7 +5577,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -5658,6 +5672,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, + {.id = ALC292_FIXUP_TPT460_DOCK, .name = "tpt460-dock"}, + {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.8.0.rc3

2 3

[alsa-devel] A tweak to the intel codec to get rid of the hiss.
by Cyrille Dunant 11 Apr '16

11 Apr '16

I apologise for the noise on the list, but I noted that on a number of HDA card, the default produces a small static noise, and I tracked it down to some channels which I believe ought to be muted. Attached is the output of alsa-info.sh, and the diff from the python hda tweak tool. This helped both for my laptop and desktop (both HDA cards, different models). Cheers! Diff for codec 1/0 (0x10ec0280): --- +++ @@ -17,17 +17,17 @@ IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Device: name="ALC3220 Analog", type="Audio", device=0 Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 - Amp-Out vals: [0x2e 0x2e] + Amp-Out vals: [0x2f 0x2f] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 @@ -66,17 +66,17 @@ rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 - Amp-In vals: [0x97 0x97] + Amp-In vals: [0x17 0x17] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 @@ -87,27 +87,27 @@ ControlAmp: chs=3, dir=1, idx=2, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=2, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 - Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] + Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x00 0x00] + Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: 0x80 0x00 Connection: 2 0x02 0x0

2 1

[alsa-devel] [PATCH 0/6] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 11 Apr '16

11 Apr '16

This series of patches adds devicetree support for the davinci McBSP audio interface.

3 20

[alsa-devel] Checkpatch warnings on switch case fallthru
by Harvey, Ryan 11 Apr '16

11 Apr '16

Hello, I receive a set of checkpatch warnings against my switch fallthru structure in readable_registers(). I¹m able to eliminate these warnings with a preceding comment, but that does messy the code. Is there a method better than switch for accomplishing this? static bool cs43130_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1: case CS43130_PWDN_CTL: case CS43130_PLL_SET_1 ... CS43130_PLL_SET_6: case CS43130_PLL_SET_7: case CS43130_PLL_SET_8: case CS43130_PLL_SET_9: case CS43130_PLL_SET_10: case CS43130_CLKOUT_CTL: case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF: case CS43130_HP_OUT_CTL_1: case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2: case CS43130_CLASS_H_CTL: case CS43130_HP_DETECT ... CS43130_HP_STATUS: case CS43130_HP_LOAD_1: case CS43130_HP_LOAD_STAT: case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5: return true; default: return false; } } 140: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_7: 141: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_8: 142: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_9: 143: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_10: 144: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_CLKOUT_CTL: 152: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_HP_OUT_CTL_1: 154: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_CLASS_H_CTL: 156: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_HP_LOAD_1: 160: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_HP_LOAD_STAT: Thanks -Ryan

4 5

[alsa-devel] [PATCH] ASoC: rockchip: i2s: configure the sdio pins' iomux mode
by Sugar Zhang 11 Apr '16

11 Apr '16

There are 3 i2s sdio pins, which iomux mode is as follows: - sdi3_sdo1 - sdi2_sdo2 - sdi1_sdo3 we need to configure these pins' iomux mode via the GRF register when use multi channel playback/capture. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- .../devicetree/bindings/sound/rockchip-i2s.txt | 5 +++ sound/soc/rockchip/rockchip_i2s.c | 39 +++++++++++++++++++++- sound/soc/rockchip/rockchip_i2s.h | 8 +++++ 3 files changed, 51 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 6e86d8a..ad72a7d 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -23,6 +23,11 @@ Required properties: - rockchip,playback-channels: max playback channels, if not set, 8 channels default. - rockchip,capture-channels: max capture channels, if not set, 2 channels default. +Required properties for controller which support multi channels playback/capture: + +- rockchip,grf: Should be phandle/offset pair. the phandle of the syscon node for GRF register, + and the offset of the GRF for control register. + Example for rk3288 I2S controller: i2s@ff890000 { diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e204..bc72780 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -11,6 +11,7 @@ */ #include <linux/module.h> +#include <linux/mfd/syscon.h> #include <linux/delay.h> #include <linux/of_gpio.h> #include <linux/clk.h> @@ -33,6 +34,8 @@ struct rk_i2s_dev { struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; + struct regmap *grf; + u32 iocfg_reg; bool is_master_mode; }; @@ -277,6 +280,29 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK, val); + if (!IS_ERR(i2s->grf)) { + regmap_read(i2s->regmap, I2S_TXCR, &val); + val &= I2S_TXCR_CSR_MASK; + + switch (val) { + case I2S_CHN_4: + val = I2S_IO_4CH_OUT_6CH_IN; + break; + case I2S_CHN_6: + val = I2S_IO_6CH_OUT_4CH_IN; + break; + case I2S_CHN_8: + val = I2S_IO_8CH_OUT_2CH_IN; + break; + default: + val = I2S_IO_2CH_OUT_8CH_IN; + break; + } + + regmap_write(i2s->grf, i2s->iocfg_reg, + I2S_IO_DIRECTION_MASK << 16 | val); + } + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, I2S_DMACR_TDL(16)); regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, @@ -478,6 +504,18 @@ static int rockchip_i2s_probe(struct platform_device *pdev) return -ENOMEM; } + i2s->dev = &pdev->dev; + + i2s->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); + if (!IS_ERR(i2s->grf)) { + ret = of_property_read_u32_index(node, "rockchip,grf", + 1, &i2s->iocfg_reg); + if (ret) { + dev_err(&pdev->dev, "Can't get iocfg_reg offset\n"); + return ret; + } + } + /* try to prepare related clocks */ i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk"); if (IS_ERR(i2s->hclk)) { @@ -517,7 +555,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; i2s->capture_dma_data.maxburst = 4; - i2s->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, i2s); pm_runtime_enable(&pdev->dev); diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index dc6e2c7..9a6aabf 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -236,4 +236,12 @@ enum { #define I2S_TXDR (0x0024) #define I2S_RXDR (0x0028) +/* io direction cfg register */ +#define I2S_IO_DIRECTION_SHIFT 11 +#define I2S_IO_DIRECTION_MASK (7 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_8CH_OUT_2CH_IN (0 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_6CH_OUT_4CH_IN (1 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_4CH_OUT_6CH_IN (3 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_2CH_OUT_8CH_IN (7 << I2S_IO_DIRECTION_SHIFT) + #endif /* _ROCKCHIP_IIS_H */ -- 1.9.1

4 4

[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 10 Apr '16

10 Apr '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: 1. Removed the probe callback as it was not used. 2. Replaced the control with tlv .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/max98371.c | 427 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 64 +++ 3 files changed, 508 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..fe7f08f --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,427 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const voltage_level_text[] = { + "9.5dB", "10.5dB", "11.5dB", "11.5dB", "12.5dB", + "13.5dB", "14.5dB", "15.5dB", "16.5dB", "17.5dB", + "18.5dB", "19.5dB", "20.5dB" +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_voltage, MAX98371_GAIN, 0, + voltage_level_text); + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const unsigned int max98371_gain_tlv[] = { + 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), + 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), + 11, 15, TLV_DB_SCALE_ITEM(0, 0, 0), +}; + +static const unsigned int max98371_noload_gain_tlv[] = { + 0, 11, TLV_DB_SCALE_ITEM(0, 100, 0), +}; + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker No-load Volume", MAX98371_GAIN, + 0, (1<<MAX98371_NO_LOAD_WIDTH)-1, 0, + max98371_noload_gain_tlv), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Voltage Level", max98371_voltage), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + pr_info("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + pr_info("%s: ratio unsupported %d", __func__, blr_clk_ratio); + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + pr_info("%s: rate unsupported %d", __func__, rate); + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", + "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Enable", + "HiFi Playback", MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", + MAX98371_GLOBAL_ENABLE, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC Enable", NULL, "DAI_OUT"}, + {"SPK_OUT", NULL, "DAC Enable"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..c75b259 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,64 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_NO_LOAD_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

3 5

[alsa-devel] Real-time audio over local network with good quality
by Ricardo Biehl 10 Apr '16

10 Apr '16

Hello guys! I'm sorry if this mail is not part of the scope of this list :-( I've come from the kernelnewbies mailing list with the same subject as following ... I'm working on a project which aims to be a microphone system that will work over local network <https://github.com/ricardobiehl/ipmic>. So I've found many difficulties in ALSA library and Linux Network. Basically, I need help to make my software get low latency and high quality audio :-) . When I said "high quality audio" I'm referring to (1channel * 16bit sample * 22050Hz) without noise. And "low latency" is about to 0.1 second or less. Any contribution with the project is welcome! Thank you a lot! -- Ricardo Biehl Pasquali

2 2

[alsa-devel] [PATCH 1/2] sound: usb: add UAC2 clock sources as mixer controls
by Daniel Mack 09 Apr '16

09 Apr '16

UAC2 specifies clock sources that optionally have validity controls. This patch exposes them as mixer controls, so they can be read (and at least in theory even be written) by userspace applications in order to make clock selection policy decisions. This implementation does nothing if the device is not UAC2 compliant, or if the clock source does not define said validity control bits. Tested with a miniDSP USBStreamer (0x2752/0x0016), Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/mixer.c | 66 ++++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 65 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757..973274b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> +#include <linux/log2.h> #include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> @@ -1378,6 +1379,68 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) + +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[100]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) + return -EINVAL; + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock source %d validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2189,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); -- 2.4.3

2 8

[alsa-devel] [PATCH v2 1/2] sound: usb: add UAC2 clock sources as mixer controls
by Daniel Mack 09 Apr '16

09 Apr '16

UAC2 specifies clock sources that optionally have validity controls. This patch exposes them as mixer controls, so they can be read (and at least in theory even be written) by userspace applications in order to make clock selection policy decisions. This implementation does nothing if the device is not UAC2 compliant, or if the clock source does not define said validity control bits. Tested with a miniDSP USBStreamer (0x2752/0x0016). Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/mixer.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757..cab6f52 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> +#include <linux/log2.h> #include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> @@ -1378,6 +1379,71 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) { + usb_audio_dbg(state->chip, + "Bogus clock source descriptor length of %d, ignoring.\n", + hdr->bLength); + return 0; + } + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s Validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock Source %d Validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2192,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); -- 2.4.3

1 1

[alsa-devel] [PATCH] ALSA: constify ct_timer_ops structures
by Julia Lawall 09 Apr '16

09 Apr '16

The ct_timer_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/pci/ctxfi/cttimer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index a5d4604..8f94534 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -49,7 +49,7 @@ struct ct_timer { spinlock_t lock; /* global timer lock (for xfitimer) */ spinlock_t list_lock; /* lock for instance list */ struct ct_atc *atc; - struct ct_timer_ops *ops; + const struct ct_timer_ops *ops; struct list_head instance_head; struct list_head running_head; unsigned int wc; /* current wallclock */ @@ -128,7 +128,7 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti) #define ct_systimer_free ct_systimer_prepare -static struct ct_timer_ops ct_systimer_ops = { +static const struct ct_timer_ops ct_systimer_ops = { .init = ct_systimer_init, .free_instance = ct_systimer_free, .prepare = ct_systimer_prepare, @@ -322,7 +322,7 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer) ct_xfitimer_irq_stop(atimer); } -static struct ct_timer_ops ct_xfitimer_ops = { +static const struct ct_timer_ops ct_xfitimer_ops = { .prepare = ct_xfitimer_prepare, .start = ct_xfitimer_start, .stop = ct_xfitimer_stop,

2 1

[alsa-devel] [PATCH] ASoC: samsung: Remove unused Odroid x2/u3 machine driver
by Sylwester Nawrocki 08 Apr '16

08 Apr '16

We have been using "simple-audio-card" for Odroid X2/U3 boards, as can be seen from sound node in arch/arm/boot/dts/ exynos4412-odroid-common.dtsi. A dedicated machine driver is not needed and it is removed in this patch. There is no dts files using "samsung,odroidx2-audio" or "samsung,odroidu3-audio" compatible strings. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,odroidx2-max98090.txt | 35 ---- sound/soc/samsung/Kconfig | 8 - sound/soc/samsung/Makefile | 2 - sound/soc/samsung/odroidx2_max98090.c | 185 --------------------- 4 files changed, 230 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt delete mode 100644 sound/soc/samsung/odroidx2_max98090.c diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt deleted file mode 100644 index 9148f72..0000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt +++ /dev/null @@ -1,35 +0,0 @@ -Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec - -Required properties: - - compatible : "samsung,odroidx2-audio" - for Odroid X2 board, - "samsung,odroidu3-audio" - for Odroid U3 board - - samsung,model : the user-visible name of this sound complex - - samsung,i2s-controller : the phandle of the I2S controller - - samsung,audio-codec : the phandle of the MAX98090 audio codec - - samsung,audio-routing : a list of the connections between audio - components; each entry is a pair of strings, the first being the - connection's sink, the second being the connection's source; - valid names for sources and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - For Odroid X2: - * Headphone Jack - * Mic Jack - * DMIC - - For Odroid U3: - * Headphone Jack - * Speakers - -Example: - -sound { - compatible = "samsung,odroidu3-audio"; - samsung,i2s-controller = <&i2s0>; - samsung,audio-codec = <&max98090>; - samsung,model = "Odroid-X2"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; -}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 78baa26..7b722b0 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -224,14 +224,6 @@ config SND_SOC_SNOW Say Y if you want to add audio support for various Snow boards based on Exynos5 series of SoCs. -config SND_SOC_ODROIDX2 - tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG && I2C - select SND_SOC_MAX98090 - select SND_SAMSUNG_I2S - help - Say Y here to enable audio support for the Odroid-X2/U3. - config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" depends on SND_SOC_SAMSUNG && I2C diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71..5d03f5c 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -43,7 +43,6 @@ snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o -snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o snd-soc-arndale-rt5631-objs := arndale_rt5631.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o @@ -69,5 +68,4 @@ obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o -obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c deleted file mode 100644 index 0421727..0000000 --- a/sound/soc/samsung/odroidx2_max98090.c +++ /dev/null @@ -1,185 +0,0 @@ -/* - * Copyright (C) 2014 Samsung Electronics Co., Ltd. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/of.h> -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include "i2s.h" - -struct odroidx2_drv_data { - const struct snd_soc_dapm_widget *dapm_widgets; - unsigned int num_dapm_widgets; -}; - -/* The I2S CDCLK output clock frequency for the MAX98090 codec */ -#define MAX98090_MCLK 19200000 - -static struct snd_soc_dai_link odroidx2_dai[]; - -static int odroidx2_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - cpu_dai = rtd->cpu_dai; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, - SND_SOC_CLOCK_IN); - - if (ret < 0 || of_find_property(odroidx2_dai[0].codec_of_node, - "clocks", NULL)) - return ret; - - /* Set the cpu DAI configuration in order to use CDCLK */ - return snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_OUT); -} - -static const struct snd_soc_dapm_widget odroidx2_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static const struct snd_soc_dapm_widget odroidu3_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speakers", NULL), -}; - -static struct snd_soc_dai_link odroidx2_dai[] = { - { - .name = "MAX98090", - .stream_name = "MAX98090 PCM", - .codec_dai_name = "HiFi", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - } -}; - -static struct snd_soc_card odroidx2 = { - .owner = THIS_MODULE, - .dai_link = odroidx2_dai, - .num_links = ARRAY_SIZE(odroidx2_dai), - .fully_routed = true, - .late_probe = odroidx2_late_probe, -}; - -static const struct odroidx2_drv_data odroidx2_drvdata = { - .dapm_widgets = odroidx2_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets), -}; - -static const struct odroidx2_drv_data odroidu3_drvdata = { - .dapm_widgets = odroidu3_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets), -}; - -static const struct of_device_id odroidx2_audio_of_match[] = { - { - .compatible = "samsung,odroidx2-audio", - .data = &odroidx2_drvdata, - }, { - .compatible = "samsung,odroidu3-audio", - .data = &odroidu3_drvdata, - }, - { }, -}; -MODULE_DEVICE_TABLE(of, odroidx2_audio_of_match); - -static int odroidx2_audio_probe(struct platform_device *pdev) -{ - struct device_node *snd_node = pdev->dev.of_node; - struct snd_soc_card *card = &odroidx2; - struct device_node *i2s_node, *codec_node; - struct odroidx2_drv_data *dd; - const struct of_device_id *of_id; - int ret; - - of_id = of_match_node(odroidx2_audio_of_match, snd_node); - dd = (struct odroidx2_drv_data *)of_id->data; - - card->num_dapm_widgets = dd->num_dapm_widgets; - card->dapm_widgets = dd->dapm_widgets; - - card->dev = &pdev->dev; - - ret = snd_soc_of_parse_card_name(card, "samsung,model"); - if (ret < 0) - return ret; - - ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); - if (ret < 0) - return ret; - - codec_node = of_parse_phandle(snd_node, "samsung,audio-codec", 0); - if (!codec_node) { - dev_err(&pdev->dev, - "Failed parsing samsung,i2s-codec property\n"); - return -EINVAL; - } - - i2s_node = of_parse_phandle(snd_node, "samsung,i2s-controller", 0); - if (!i2s_node) { - dev_err(&pdev->dev, - "Failed parsing samsung,i2s-controller property\n"); - ret = -EINVAL; - goto err_put_codec_n; - } - - odroidx2_dai[0].codec_of_node = codec_node; - odroidx2_dai[0].cpu_of_node = i2s_node; - odroidx2_dai[0].platform_of_node = i2s_node; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - goto err_put_i2s_n; - } - return 0; - -err_put_i2s_n: - of_node_put(i2s_node); -err_put_codec_n: - of_node_put(codec_node); - return ret; -} - -static int odroidx2_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - of_node_put(odroidx2_dai[0].cpu_of_node); - of_node_put(odroidx2_dai[0].codec_of_node); - - return 0; -} - -static struct platform_driver odroidx2_audio_driver = { - .driver = { - .name = "odroidx2-audio", - .of_match_table = odroidx2_audio_of_match, - .pm = &snd_soc_pm_ops, - }, - .probe = odroidx2_audio_probe, - .remove = odroidx2_audio_remove, -}; -module_platform_driver(odroidx2_audio_driver); - -MODULE_AUTHOR("Chen Zhen <zhen1.chen(a)samsung.com>"); -MODULE_AUTHOR("Sylwester Nawrocki <s.nawrocki(a)samsung.com>"); -MODULE_DESCRIPTION("ALSA SoC Odroid X2/U3 Audio Support"); -MODULE_LICENSE("GPL v2"); -- 1.9.1

1 0

[alsa-devel] [PATCH 1/3] ASoC: samsung: Remove unused "samsung-i2sv4" platform_device_id entry
by Sylwester Nawrocki 08 Apr '16

08 Apr '16

"samsung-i2sv4" identifier was previously used for the I2S device of the S5PV210 SoCs, it can be removed now when s5pv210 is a dt-only platform. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 70a2559..ededac9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1492,9 +1492,6 @@ static const struct platform_device_id samsung_i2s_driver_ids[] = { }, { .name = "samsung-i2s-sec", .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, - }, { - .name = "samsung-i2sv4", - .driver_data = (kernel_ulong_t)&i2sv5_dai_type, }, {}, }; -- 1.9.1

1 2

[alsa-devel] [PATCH] ASoC: arizona: Do not create OUT4R widget for CS47L24/WM1831
by Richard Fitzgerald 08 Apr '16

08 Apr '16

The CS47L24 and WM1831 codecs only use the OUT4L widget so we can skip creation of the OUT4R widget. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 92d22a0..d8a6823 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); -- 1.9.1

1 0

[alsa-devel] [PATCH v2 0/6] Propagate errors out from compressed streams
by Charles Keepax 08 Apr '16

08 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 v1: - Based on a recommendation from Vinod, rather than handling errors from pointer requests, add a new call that allows us to inform the compress core that the stream has gone bad. Then return errors from the avail and poll callbacks if the stream is bad. Thanks, Charles Charles Keepax (6): ASoC: compress: Pass error out of soc_compr_pointer ALSA: compress: Add function to indicate the stream has gone bad ALSA: compress: Replace complex if statement with switch ASoC: wm_adsp: Factor out fetching of stream errors from the DSP ASoC: wm_adsp: Improve DSP error handling ASoC: wm_adsp: Use new snd_compr_stop_xrun to signal stream failure include/sound/compress_driver.h | 3 ++ sound/core/compress_offload.c | 67 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 43 +++++++++++++++++++------- sound/soc/soc-compress.c | 5 +-- 4 files changed, 101 insertions(+), 17 deletions(-) -- 2.1.4

4 13

[alsa-devel] [PATCH] [linux-next]:ALSA: Fix a typo in timestamping.txt
by Masanari Iida 08 Apr '16

08 Apr '16

This patch fix a spelling typo found in Documentation/sound/alsa/timestamping.txt Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- Documentation/sound/alsa/timestamping.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt index 0b191a23f534..1b6473f393a8 100644 --- a/Documentation/sound/alsa/timestamping.txt +++ b/Documentation/sound/alsa/timestamping.txt @@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an interpolation of the results In some hardware-specific configuration, the system timestamp is -latched by a low-level audio subsytem, and the information provided +latched by a low-level audio subsystem, and the information provided back to the driver. Due to potential delays in the communication with the hardware, there is a risk of misalignment with the avail and delay information. To make sure applications are not confused, a -- 2.8.0

2 1

[alsa-devel] [PATCH v3 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 07 Apr '16

07 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. History: v2: add check on string length, use strtol() to get hex value, and fix memory leak. v3: Fix overflow on uuid string copy. Use strtoul() to get unsigned vendor values and check range, tested on 32 & 64 bit systems. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 +++- include/topology.h | 79 +++++++- src/topology/data.c | 503 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 702 insertions(+), 8 deletions(-) -- 2.5.0

3 10

[alsa-devel] [PATCH] pcm_plugin: fix appl pointer not correct when mmap_commit() return error
by Shengjiu Wang 07 Apr '16

07 Apr '16

When snd_pcm_mmap_commit() return error, the appl pointer is also updated. which cause the avail_update()'s result wrong. This patch move the snd_pcm_mmap_appl_forward() to the place when snd_pcm_mmap_commit() is successfully returned. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_plugin.c | 48 ++++++++++++++++++++++++++++++++---------------- 1 file changed, 32 insertions(+), 16 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index d007e8c..940491d 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -279,18 +279,22 @@ static snd_pcm_sframes_t snd_pcm_plugin_write_areas(snd_pcm_t *pcm, return -EPIPE; } snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_write(pcm, slave_areas, slave_offset + result, slave_frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - snd_atomic_write_end(&plugin->watom); - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); offset += frames; xfer += frames; size -= frames; @@ -325,19 +329,23 @@ static snd_pcm_sframes_t snd_pcm_plugin_read_areas(snd_pcm_t *pcm, return -EPIPE; } snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_read(slave, areas, offset, frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - snd_atomic_write_end(&plugin->watom); - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); offset += frames; xfer += frames; size -= frames; @@ -423,19 +431,23 @@ snd_pcm_plugin_mmap_commit(snd_pcm_t *pcm, frames = plugin->write(pcm, areas, appl_offset, frames, slave_areas, slave_offset, &slave_frames); snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); - snd_atomic_write_end(&plugin->watom); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_write(pcm, slave_areas, slave_offset + result, slave_frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? xfer : res; + } frames -= res; } - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); if (frames == cont) appl_offset = 0; else @@ -490,19 +502,23 @@ static snd_pcm_sframes_t snd_pcm_plugin_avail_update(snd_pcm_t *pcm) frames = (plugin->read)(pcm, areas, hw_offset, frames, slave_areas, slave_offset, &slave_frames); snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_hw_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); - snd_atomic_write_end(&plugin->watom); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_read(slave, areas, hw_offset, frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_hw_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); if (frames == cont) hw_offset = 0; else -- 1.9.1

2 1

[alsa-devel] Synchonize video with audio playback
by Li, Steve 07 Apr '16

07 Apr '16

Hi, I am new to ALSA. I am trying to implement video and audio playback in my app. I am able to find a lot of sample code for audio playback, but I am wondering what is the best way to synchronize video playback with audio. What kind of timing information can I query through the ALSA interface in terms of the audio playback? Your help is appreciated. Thanks Steve ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

2 1

[alsa-devel] [PATCH V7 0/2] ASoC: Add core API to register and cleanup DMI names for card
by han.lu＠intel.com 07 Apr '16

07 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, for user space utils such as PA and UCM to distinguish different products. 1. Add core APIs to register and cleanup DMI names for card. 2. Apply the APIs to bytcr-rt5640 driver. changes on V7: 1. Remove inconsistent API description changes on V6: 1. Use dynamic allocate and cleanup for card long name 2. Remove unneccessary arguments to simplify the API changes on V5: 1. Use independent space to store card long_name, to avoid irrelavant info sharing from card component 2. Use letter ';' instead of ':' to separate strings in long name, in case name strings may also contain ':' and confuse user 3. Fix error that vendor name and firmware name were not optional changes on V4: 1. Replace kmalloc() and snprintf() with ksaprintf() to simplify code changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering and cleaning up DMI card names ASoC: bytcr-rt5640: register DMI names for card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 18 ++++++++ sound/soc/soc-core.c | 85 +++++++++++++++++++++++++++++++++++ 3 files changed, 106 insertions(+) -- 2.5.0

5 8

[alsa-devel] [patch] ASoC: Intel: sst: fix a loop timeout in sst_hsw_stream_reset()
by Dan Carpenter 07 Apr '16

07 Apr '16

In the original code we ended the loop with tries set to -1 instead of zero. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Not tested. diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index ac60f13..9156522 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1345,7 +1345,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; /* wait for pause to complete before we reset the stream */ - while (stream->running && tries--) + while (stream->running && --tries) msleep(1); if (!tries) { dev_err(hsw->dev, "error: reset stream %d still running\n",

3 2

[alsa-devel] CircuitCo AudioCape and 4.4.6-bone-rt-r7 kernel, not quite there
by Rick Mann 07 Apr '16

07 Apr '16

I'm thinking my problems are more ALSA-related than Beaglebone-specific. I wonder if anyone can help. With kernel 4.4.6-bone-rt-r7, and the CircuitCo AudioCape, I get a sound card in the system, but speaker-test reports errors (Unable to set hw params for playback: Invalid argument). I may have issues in various ALSA config settings; I'm hoping someone here can help me by showing me what to look at and change. The CODEC is a tlv320aic3104. Thanks! ----------- # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: B [AudioCape Rev B], device 0: davinci-mcasp.0-tlv320aic3x-hifi tlv320aic3x-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ----------- # speaker-test speaker-test 1.0.28 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 16 to 32768 Period size range from 8 to 16384 Using max buffer size 32768 Periods = 4 Unable to set hw params for playback: Invalid argument Setting of hwparams failed: Invalid argument ----- More info: My DTB: http://pastebin.com/Wja7ZQRk and DTBO: https://github.com/RobertCNelson/bb.org-overlays/blob/master/src/arm/BB-BON… Results of speaker-test, aplay, etc.: http://pastebin.com/mMPbReAV One possibility, ALSA is still not configured correctly for this cape, but I wonder if anyone has the right configuration for it? Thanks! -- Rick Mann rmann(a)latencyzero.com

1 0

[alsa-devel] [PATCH] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 06 Apr '16

06 Apr '16

This patch adds second I2S for rt5650 codec on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 55 ++++++++++++++++++-- 2 files changed, 56 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..0046926 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: +- mediatek,rt5650_i2s: I2S mode of rt5650 + 0: using I2S1 on rt5650 for record + 1: using I2S2 on rt5650 for record + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + mediatek,rt5650_i2s = <0>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..ab4c806 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -23,6 +23,20 @@ #define MCLK_FOR_CODECS 12288000 +struct mt8173_rt5650_platform_data { + unsigned int rt5650_i2s; + /* 0 = I2S1; 1 = I2S2 */ +}; + +static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { + .rt5650_i2s = 0, +}; + +enum mt8173_rt5650_i2s { + MT8173_RT5650_I2S1 = 0, + MT8173_RT5650_I2S2, +}; + static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), @@ -87,10 +101,25 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; int ret; - rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, - RT5645_CLK_SEL_I2S1_ASRC); + switch (mt8173_rt5650_priv.rt5650_i2s) { + case MT8173_RT5650_I2S1: + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + break; + case MT8173_RT5650_I2S2: + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + break; + default: + break; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -112,6 +141,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { .dai_name = "rt5645-aif1", }, + { + .dai_name = "rt5645-aif2", + }, }; enum { @@ -149,7 +181,7 @@ 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, @@ -199,6 +231,19 @@ 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, "mediatek,rt5650_i2s")) { + ret = device_property_read_u32(&pdev->dev, + "mediatek,rt5650_i2s", + &mt8173_rt5650_priv.rt5650_i2s); + if (ret) { + dev_err(&pdev->dev, + "%s snd_soc_register_card fail %d\n", + __func__, ret); + } + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 7

[alsa-devel] [PATCH v2] ASoC: rt5514: add rt5514 SPI driver
by Oder Chiou 06 Apr '16

06 Apr '16

The device has multiple control interfaces, I2C and SPI. The I2C interface mainly controls the register settings of codec. The SPI interface is in order to provide the high speed transmission of data. For example, high bandwidth memory read/write of DSP. The patch adds the rt5514 SPI driver for loading the firmware of DSP and retrieving the voice data from DSP after the system is waked up by specific voice. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5514-spi.c | 458 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514-spi.h | 38 ++++ sound/soc/codecs/rt5514.c | 249 +++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 4 + 6 files changed, 754 insertions(+) create mode 100644 sound/soc/codecs/rt5514-spi.c create mode 100644 sound/soc/codecs/rt5514-spi.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f99b11..306c4e3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -627,6 +627,9 @@ config SND_SOC_RT298 config SND_SOC_RT5514 tristate +config SND_SOC_RT5514_SPI + tristate + config SND_SOC_RT5616 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712..501bf43 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -97,6 +97,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o +snd-soc-rt5514-spi-objs := rt5514-spi.o snd-soc-rt5616-objs := rt5616.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -306,6 +307,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o +obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c new file mode 100644 index 0000000..631865d --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.c @@ -0,0 +1,458 @@ +/* + * rt5514-spi.c -- RT5514 SPI driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/input.h> +#include <linux/spi/spi.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/sched.h> +#include <linux/kthread.h> +#include <linux/uaccess.h> +#include <linux/miscdevice.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_qos.h> +#include <linux/sysfs.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5514-spi.h" + +static struct spi_device *rt5514_spi; + +struct rt5514_dsp { + struct device *dev; + struct delayed_work copy_work; + struct mutex dma_lock; + struct snd_pcm_substream *substream; + unsigned int buf_base, buf_limit, buf_rp; + size_t buf_size; + size_t dma_offset; + size_t dsp_offset; +}; + +static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = 0x20000 / 8, + .periods_min = 8, + .periods_max = 8, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 0x20000, +}; + +static struct snd_soc_dai_driver rt5514_spi_dai = { + .name = "rt5514-dsp-cpu-dai", + .id = 0, + .capture = { + .stream_name = "DSP Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static void rt5514_spi_copy_work(struct work_struct *work) +{ + struct rt5514_dsp *rt5514_dsp = + container_of(work, struct rt5514_dsp, copy_work.work); + struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; + size_t period_bytes, truncated_bytes = 0; + + mutex_lock(&rt5514_dsp->dma_lock); + if (!rt5514_dsp->substream) { + dev_err(rt5514_dsp->dev, "No pcm substream\n"); + goto done; + } + + period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + + if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) + period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset; + + if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + period_bytes); + + if (rt5514_dsp->buf_rp + period_bytes == rt5514_dsp->buf_limit) + rt5514_dsp->buf_rp = rt5514_dsp->buf_base; + else + rt5514_dsp->buf_rp += period_bytes; + } else { + truncated_bytes = rt5514_dsp->buf_limit - rt5514_dsp->buf_rp; + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + truncated_bytes); + + rt5514_spi_burst_read(rt5514_dsp->buf_base, + runtime->dma_area + rt5514_dsp->dma_offset + + truncated_bytes, period_bytes - truncated_bytes); + + rt5514_dsp->buf_rp = rt5514_dsp->buf_base + + period_bytes - truncated_bytes; + } + + rt5514_dsp->dma_offset += period_bytes; + if (rt5514_dsp->dma_offset >= runtime->dma_bytes) + rt5514_dsp->dma_offset = 0; + + rt5514_dsp->dsp_offset += period_bytes; + + snd_pcm_period_elapsed(rt5514_dsp->substream); + + if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 5); +done: + mutex_unlock(&rt5514_dsp->dma_lock); +} + +/* PCM for streaming audio from the DSP buffer */ +static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &rt5514_spi_pcm_hardware); + + return 0; +} + +static int rt5514_spi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + int ret; + + mutex_lock(&rt5514_dsp->dma_lock); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + rt5514_dsp->substream = substream; + mutex_unlock(&rt5514_dsp->dma_lock); + + return ret; +} + +static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + mutex_lock(&rt5514_dsp->dma_lock); + cancel_delayed_work_sync(&rt5514_dsp->copy_work); + rt5514_dsp->substream = NULL; + mutex_unlock(&rt5514_dsp->dma_lock); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int rt5514_spi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + u8 buf[8]; + + rt5514_dsp->dma_offset = 0; + rt5514_dsp->dsp_offset = 0; + + /** + * The address area x1800XXXX is the register address, and it cannot + * support spi burst read perfectly. So we use the spi burst read + * individually to make sure the data correctly. + */ + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + return 0; +} + +static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); + } + + return 0; +} + +static snd_pcm_uframes_t rt5514_spi_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + return bytes_to_frames(runtime, rt5514_dsp->dma_offset); +} + +static struct snd_pcm_ops rt5514_spi_pcm_ops = { + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .trigger = rt5514_spi_trigger, + .prepare = rt5514_spi_prepare, + .pointer = rt5514_spi_pcm_pointer, + .mmap = snd_pcm_lib_mmap_vmalloc, + .page = snd_pcm_lib_get_vmalloc_page, +}; + +static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) +{ + struct rt5514_dsp *rt5514_dsp; + + rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), + GFP_KERNEL); + + rt5514_dsp->dev = &rt5514_spi->dev; + mutex_init(&rt5514_dsp->dma_lock); + INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); + snd_soc_platform_set_drvdata(platform, rt5514_dsp); + + return 0; +} + +static struct snd_soc_platform_driver rt5514_spi_platform = { + .probe = rt5514_spi_pcm_probe, + .ops = &rt5514_spi_pcm_ops, +}; + +static const struct snd_soc_component_driver rt5514_spi_dai_component = { + .name = "rt5514-spi-dai", +}; + +/** + * rt5514_spi_burst_read - Read data from SPI by rt5514 address. + * @addr: Start address. + * @rxbuf: Data Buffer for reading. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_READ; + int status; + u8 write_buf[8]; + unsigned int i, end, offset = 0; + + struct spi_message message; + struct spi_transfer x[3]; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + spi_message_init(&message); + memset(x, 0, sizeof(x)); + + x[0].len = 5; + x[0].tx_buf = write_buf; + spi_message_add_tail(&x[0], &message); + + x[1].len = 4; + x[1].tx_buf = write_buf; + spi_message_add_tail(&x[1], &message); + + x[2].len = end; + x[2].rx_buf = rxbuf + offset; + spi_message_add_tail(&x[2], &message); + + status = spi_sync(rt5514_spi, &message); + + if (status) + return false; + + offset += RT5514_SPI_BUF_LEN; + } + + for (i = 0; i < len; i += 8) { + write_buf[0] = rxbuf[i + 0]; + write_buf[1] = rxbuf[i + 1]; + write_buf[2] = rxbuf[i + 2]; + write_buf[3] = rxbuf[i + 3]; + write_buf[4] = rxbuf[i + 4]; + write_buf[5] = rxbuf[i + 5]; + write_buf[6] = rxbuf[i + 6]; + write_buf[7] = rxbuf[i + 7]; + + rxbuf[i + 0] = write_buf[7]; + rxbuf[i + 1] = write_buf[6]; + rxbuf[i + 2] = write_buf[5]; + rxbuf[i + 3] = write_buf[4]; + rxbuf[i + 4] = write_buf[3]; + rxbuf[i + 5] = write_buf[2]; + rxbuf[i + 6] = write_buf[1]; + rxbuf[i + 7] = write_buf[0]; + } + + return true; +} + +/** + * rt5514_spi_burst_write - Write data to SPI by rt5514 address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int i, end, offset = 0; + + write_buf = kmalloc(RT5514_SPI_BUF_LEN + 6, GFP_KERNEL); + + if (write_buf == NULL) + return -ENOMEM; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + for (i = 0; i < end; i += 8) { + write_buf[i + 12] = txbuf[offset + i + 0]; + write_buf[i + 11] = txbuf[offset + i + 1]; + write_buf[i + 10] = txbuf[offset + i + 2]; + write_buf[i + 9] = txbuf[offset + i + 3]; + write_buf[i + 8] = txbuf[offset + i + 4]; + write_buf[i + 7] = txbuf[offset + i + 5]; + write_buf[i + 6] = txbuf[offset + i + 6]; + write_buf[i + 5] = txbuf[offset + i + 7]; + } + + write_buf[end + 5] = spi_cmd; + + spi_write(rt5514_spi, write_buf, end + 6); + + offset += RT5514_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5514_spi_burst_write); + +static int rt5514_spi_probe(struct spi_device *spi) +{ + int ret; + + rt5514_spi = spi; + + ret = snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register platform.\n"); + goto err_plat; + } + + ret = snd_soc_register_component(&spi->dev, &rt5514_spi_dai_component, + &rt5514_spi_dai, 1); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register component.\n"); + goto err_comp; + } + + return 0; +err_comp: + snd_soc_unregister_platform(&spi->dev); +err_plat: + + return 0; +} + +static int rt5514_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_component(&spi->dev); + snd_soc_unregister_platform(&spi->dev); + + return 0; +} + +static const struct of_device_id rt5514_of_match[] = { + { .compatible = "realtek,rt5514", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5514_of_match); + +static struct spi_driver rt5514_spi_driver = { + .driver = { + .name = "rt5514", + .of_match_table = of_match_ptr(rt5514_of_match), + }, + .probe = rt5514_spi_probe, + .remove = rt5514_spi_remove, +}; +module_spi_driver(rt5514_spi_driver); + +MODULE_DESCRIPTION("RT5514 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h new file mode 100644 index 0000000..f69b1cd --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.h @@ -0,0 +1,38 @@ +/* + * rt5514-spi.h -- RT5514 driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5514_SPI_H__ +#define __RT5514_SPI_H__ + +/** + * RT5514_SPI_BUF_LEN is the buffer size of SPI master controller. +*/ +#define RT5514_SPI_BUF_LEN 240 + +#define RT5514_BUFFER_VOICE_BASE 0x18001034 +#define RT5514_BUFFER_VOICE_LIMIT 0x18001038 +#define RT5514_BUFFER_VOICE_RP 0x1800103c +#define RT5514_BUFFER_VOICE_SIZE 0x18001040 + +/* SPI Command */ +enum { + RT5514_SPI_CMD_16_READ = 0, + RT5514_SPI_CMD_16_WRITE, + RT5514_SPI_CMD_32_READ, + RT5514_SPI_CMD_32_WRITE, + RT5514_SPI_CMD_BURST_READ, + RT5514_SPI_CMD_BURST_WRITE, +}; + +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len); +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5514_SPI_H__ */ diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 879bf60..517242a 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -30,6 +30,9 @@ #include "rl6231.h" #include "rt5514.h" +#if defined(CONFIG_SND_SOC_RT5514_SPI) +#include "rt5514-spi.h" +#endif static const struct reg_sequence rt5514_i2c_patch[] = { {0x1800101c, 0x00000000}, @@ -110,6 +113,159 @@ static const struct reg_default rt5514_reg[] = { {RT5514_VENDOR_ID2, 0x10ec5514}, }; +static void rt5514_enable_dsp_prepare(struct rt5514_priv *rt5514) +{ + /* Reset */ + regmap_write(rt5514->i2c_regmap, 0x18002000, 0x000010ec); + /* LDO_I_limit */ + regmap_write(rt5514->i2c_regmap, 0x18002200, 0x00028604); + /* (for reset DSP) mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x0005514b); + /* (for reset DSP) mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* DMIC_OUT1/2=HiZ */ + regmap_write(rt5514->i2c_regmap, 0x180020a0, 0x00000000); + /* PIN config */ + regmap_write(rt5514->i2c_regmap, 0x18002070, 0x00000040); + /* pll3(QN)=RCOSC*(10+2) */ + regmap_write(rt5514->i2c_regmap, 0x18002240, 0x0000000a); + /* pll3 source=RCOSC, fsi=rt_clk */ + regmap_write(rt5514->i2c_regmap, 0x18002100, 0x0000000b); + /* PU RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* PD ADC1/2 */ + regmap_write(rt5514->i2c_regmap, 0x18002008, 0x00220000); + /* DSP clk source=pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* 256fs=/4,DMIC_CLK_OUT=/16(disable ad2) */ + regmap_write(rt5514->i2c_regmap, 0x18002104, 0x10023541); + /* clk_sys source=mux_out */ + regmap_write(rt5514->i2c_regmap, 0x18002108, 0x00000000); + /* DSP clk= 15M*(20+1)/32 =10M */ + regmap_write(rt5514->i2c_regmap, 0x18001100, 0x00000214); + /* DB, pointer */ + regmap_write(rt5514->i2c_regmap, 0x18002148, 0x80000000); + /* DB, pop=4x */ + regmap_write(rt5514->i2c_regmap, 0x18002140, 0x3fff00fa); + /* DB, pointer */ + regmap_write(rt5514->i2c_regmap, 0x18002148, 0x00000000); + /* DFLL reset */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0x00220012); + /* DFLL, set m/n */ + regmap_write(rt5514->i2c_regmap, 0x18002110, 0x000101f4); + /* DFLL,reset DFLL */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0x80220012); + /* DFLL */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0xc0220012); + /* (I2S) i2s format, TDM 4ch */ + regmap_write(rt5514->i2c_regmap, 0x18002010, 0x10000772); + /* (I2S) source sel; tdm_0=ad0, tdm_1=ad1 */ + regmap_write(rt5514->i2c_regmap, 0x180020ac, 0x44000eee); + /* (ad0) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x18002190, 0x0002042f); + /* (ad0) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x18002194, 0x0002042f); + /* (ad0) DMIC-IN1 L/R select */ + regmap_write(rt5514->i2c_regmap, 0x18002198, 0x10000362); + /* (ad1) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x180021a0, 0x0002042f); + /* (ad1) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x180021a4, 0x0002042f); + /* (ad1) DMIC-IN2 L/R select */ + regmap_write(rt5514->i2c_regmap, 0x180021a8, 0x10000362); + /* (ad2) gain=24dB for WOV */ + regmap_write(rt5514->i2c_regmap, 0x180020d0, 0x00008a2f); + /* dsp clk auto switch */ + regmap_write(rt5514->i2c_regmap, 0x18001114, 0x00000001); + /* reduce DSP power */ + regmap_write(rt5514->i2c_regmap, 0x18001118, 0x00000001); + /* UART clk=off */ + regmap_write(rt5514->i2c_regmap, 0x18001104, 0x00000003); + /* (pitch VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x1800201c, 0x69f32067); + /* (pitch VAD)fix2 */ + regmap_write(rt5514->i2c_regmap, 0x18002020, 0x50d500a5); + /* (pitch VAD)fix3 */ + regmap_write(rt5514->i2c_regmap, 0x18002024, 0x000a0206); + /* (pitch VAD)fix4 */ + regmap_write(rt5514->i2c_regmap, 0x18002028, 0x01800114); + /* (hello VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x18002038, 0x00100010); + /* (ok VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x1800204c, 0x000503c8); + /* (ok VAD)fix2 */ + regmap_write(rt5514->i2c_regmap, 0x18002050, 0x001a0308); + /* (ok VAD)fix3 */ + regmap_write(rt5514->i2c_regmap, 0x18002054, 0x50020502); + /* (ok VAD)fix4 */ + regmap_write(rt5514->i2c_regmap, 0x18002058, 0x50000d18); + /* (ok VAD)fix5 */ + regmap_write(rt5514->i2c_regmap, 0x1800205c, 0x640c0b14); + /* (ok VAD)fix6 */ + regmap_write(rt5514->i2c_regmap, 0x18002060, 0x00100001); + /* (FW) SENSORY_SVSCORE(for UDT+SID) */ + regmap_write(rt5514->i2c_regmap, 0x18002fa4, 0x00000000); + /* (FW) SENSORY_THRS(for UDT+SID) */ + regmap_write(rt5514->i2c_regmap, 0x18002fa8, 0x00000000); + /* (FW) DLY_BUF_LTC_OFFSET (for ok/hello VAD) */ + regmap_write(rt5514->i2c_regmap, 0x18002fbc, 0x00000000); +} + +static void rt5514_enable_dsp(struct rt5514_priv *rt5514) +{ + /* (FW) DRIVER_CTRL0: (1)VAD timeout[7:0]: unit = 0.1s + * (2)buffer mode[15:14]: + * 0 = key phrase+voice command + * 1 = voice command + * 2 = key phrase + */ + regmap_write(rt5514->i2c_regmap, 0x18001028, 0x0000000a); + /* PU RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* PD ADC1/2 */ + regmap_write(rt5514->i2c_regmap, 0x18002008, 0x00220000); + /* dsp stop */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* CLK, 256fs=/4,DMIC_CLK_OUT=/16(enable ad2) */ + regmap_write(rt5514->i2c_regmap, 0x18002104, 0x14023541); + /* CLK, clk_sys=mux_out/1 */ + regmap_write(rt5514->i2c_regmap, 0x18002108, 0x00000000); + /* (PATH) DMIC_IN1(ri)->ad2, ad2(db_PCM)->IB2 */ + regmap_write(rt5514->i2c_regmap, 0x180020a4, 0x00808002); + /* DSP clk source=pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* (opt)VAD, clr and enable pitch/hello VAD + * (0x800007d2 for ok VAD, 0x800007d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x800007d3); + /* (opt)VAD, clr and enable ok VAD + * (0x80000003 for ok VAD, 0x80000002 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x80000002); + /* (opt)VAD type sel(..90h:hello, ..80h:pitch, ..A0h:ok) */ + regmap_write(rt5514->i2c_regmap, 0x1800206c, 0x80000080); + /* dsp run */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); + /* clear IRQ */ + regmap_write(rt5514->i2c_regmap, 0x18002e04, 0x00000000); + /* (opt)VAD, release and enable pitch/hello VAD + * (0x7d2 for ok VAD, 0x7d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x000007d3); + /* (opt)VAD, release and enable ok VAD + * (0x3 for ok VAD, 0x2 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x00000002); + /* (opt)VAD, release and enable pitch/hello VAD + * (0x7d2 for ok VAD, 0x7d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x000007d3); + /* (opt)VAD, release and enable ok VAD + * (0x3 for ok VAD, 0x2 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x00000002); +} + static bool rt5514_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { @@ -248,6 +404,64 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static int rt5514_dsp_voice_wake_up_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt5514->dsp_enabled; + + return 0; +} + +static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + const struct firmware *fw = NULL; + + if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) + return 0; + + rt5514->dsp_enabled = ucontrol->value.integer.value[0]; + + if (rt5514->dsp_enabled) { + rt5514_enable_dsp_prepare(rt5514); + + request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ff60000, fw->data, + ((fw->size/8)+1)*8); +#endif + release_firmware(fw); + fw = NULL; + } + + request_firmware(&fw, RT5514_FIRMWARE2, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ffc0000, fw->data, + ((fw->size/8)+1)*8); +#endif + release_firmware(fw); + fw = NULL; + } + + rt5514_enable_dsp(rt5514); + } else { + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + + return 0; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -257,6 +471,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_R_TLV("ADC2 Capture Volume", RT5514_DOWNFILTER1_CTRL1, RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, adc_vol_tlv), + SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, + rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), }; /* ADC Mixer*/ @@ -799,6 +1015,38 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5514_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (snd_soc_codec_get_bias_level(codec) == + SND_SOC_BIAS_STANDBY) { + /** + * If the DSP is enabled in start of recording, the + * DSP should be disabled, and sync back to normal + * recording settings to make sure recording + * properly. + */ + if (rt5514->dsp_enabled) { + rt5514->dsp_enabled = 0; + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + } + break; + + default: + break; + } + + return 0; +} + static int rt5514_probe(struct snd_soc_codec *codec) { struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); @@ -857,6 +1105,7 @@ struct snd_soc_dai_driver rt5514_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_rt5514 = { .probe = rt5514_probe, + .set_bias_level = rt5514_set_bias_level, .idle_bias_off = true, .controls = rt5514_snd_controls, .num_controls = ARRAY_SIZE(rt5514_snd_controls), diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 6ad8a61..6e89e7d 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -225,6 +225,9 @@ #define RT5514_PLL_INP_MAX 40000000 #define RT5514_PLL_INP_MIN 256000 +#define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" +#define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" + /* System Clock Source */ enum { RT5514_SCLK_S_MCLK, @@ -247,6 +250,7 @@ struct rt5514_priv { int pll_src; int pll_in; int pll_out; + int dsp_enabled; }; #endif /* __RT5514_H__ */ -- 1.8.1.1.439.g50a6b54

2 7

[alsa-devel] [PATCH v2 0/2] conf: topology: Add the conf for SKL-rt286
by Vinod Koul 06 Apr '16

06 Apr '16

From: Shreyas NC <shreyas.nc(a)intel.com> The SKL RT286 I2S topology conf is provided here. The first patch adds that along with binary data for modules. The Second patch provides source of those binayy blobs and allows them to be recompiled. Shreyas NC (2): conf: topology: Add Skylake i2s conf conf: topology: Generate Private data binary blobs configure.ac | 1 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 4 + src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/data/Makefile.am | 4 + src/conf/topology/sklrt286/data/README | 2 + src/conf/topology/sklrt286/data/compile_local | 16 + src/conf/topology/sklrt286/data/configure.ac | 30 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_data_local.h | 115 ++ .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 29 files changed, 2420 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin create mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/README create mode 100755 src/conf/topology/sklrt286/data/compile_local create mode 100644 src/conf/topology/sklrt286/data/configure.ac create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data_local.h create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin create mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin create mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin create mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 2.4.11

4 10

[alsa-devel] [PATCH] Add a quirk for Plantronics BT300
by Dennis Kadioglu 06 Apr '16

06 Apr '16

Plantronics BT300 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x1". This patch adds the USB ID of the BT300 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk(a)post.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6178bb5..a97c949 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1134,6 +1134,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ ---

2 1

[alsa-devel] Help with wm8731 support for the Utilite (imx6q)
by Jonathan Bennett 06 Apr '16

06 Apr '16

Hey, I'm working with a compulab utilite, trying to get everything working with 3.17, with the eventual goal of getting better support for it upstream. Currently fighting the sound. It's a imx6q talking to a wm8731 over i2c/i2s. I am booting Fedora 21 arm Alpha, with my own dtb ( http://pastebin.com/4a4PBKwx) and driver (http://pastebin.com/1CsesuVV) It shows up in aplay -l, but trying to play anything to it gives me a div by 0 in a driver: [<c04f27e4>] (Ldiv0) from [<bf50512c>] (fsl_ssi_hw_params+0x194/0x364 [snd_soc_fsl_ssi]) This div0 happens in sound/soc/fsl/fsl_ssi.c, in the function fsl_ssi_set_bclk. My driver file is an almost direct copy of what compulab shipped with a 3.10 kernel. This is my first attempt at kernel hacking, so any help would be appreciated. ~Jonathan Bennett

4 19

[alsa-devel] simple-audio-card and external dynamic clock
by Emmanuel Fusté 06 Apr '16

06 Apr '16

Hello, I m very new to ASoC (and not native english speaker) so be indulgent ;-) The context : am335x-boneblack. I want to drive simple I2S targets. With the ongoing developments, and recent patches posted here, this is a very simple job for the simple-audio-card machine driver even with fixed external master clock. I want to go further using a programmable external clock (si570) which is not very complicated thanks to the CCF. But now I want to use the dynamic nature of this external master clock (through CCF) to be able to generate 44.1khz AND 48 multiples of fs which is not natively possible on the BBB because of integer fs scaling only and/or no dedicated audio PLL. I know that the same could be achieved on the BBB with switching between internal clock (24mhz) and external one (24.576mhz) gated by GPIO1_27, but this is another story. Which direction is the right one ? - dedicated machine driver ? - or something more generic / reusable implemented in the simple-audio-card drivers through helpers routines ? - or something else ? Than you. Emmanuel.

2 4

[alsa-devel] [PATCH] Add a quirk for Plantronics BT300
by denk 06 Apr '16

06 Apr '16

Plantronics BT300 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x1". This patch adds the USB ID of the BT300 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk(a)post.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6178bb5..a97c949 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1134,6 +1134,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ ---

1 0

[alsa-devel] [PATCH V6 0/2] ASoC: Add core API to register and cleanup DMI names for card
by han.lu＠intel.com 06 Apr '16

06 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, for user space utils such as PA and UCM to distinguish different products. 1. Add core APIs to register and cleanup DMI names for card. 2. Apply the APIs to bytcr-rt5640 driver. changes on V6: 1. Use dynamic allocate and cleanup for card long name 2. Remove unneccessary arguments to simplify the API changes on V5: 1. Use independent space to store card long_name, to avoid irrelavant info sharing from card component 2. Use letter ';' instead of ':' to separate strings in long name, in case name strings may also contain ':' and confuse user 3. Fix error that vendor name and firmware name were not optional changes on V4: 1. Replace kmalloc() and snprintf() with ksaprintf() to simplify code changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering and cleaning up DMI card names ASoC: bytcr-rt5640: register DMI names for card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 18 ++++++++ sound/soc/soc-core.c | 87 +++++++++++++++++++++++++++++++++++ 3 files changed, 108 insertions(+) -- 2.5.0

3 4

[alsa-devel] [PATCH V5 0/2] ASoC: Add core API to register DMI names to card
by han.lu＠intel.com 06 Apr '16

06 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, for user space utils such as PA and UCM to distinguish different products. 1. Add a core API to register DMI names to card. 2. Apply the API to bytcr-rt5640 driver. changes on V5: 1. Use independent space to store card long_name, to avoid irrelavant info sharing from card component 2. Use letter ';' instead of ':' to separate strings in long name, in case name strings may also contain ':' and confusing user 3. Fix error that vendor name and firmware name were not optional changes on V4: 1. Replace kmalloc() and snprintf() with ksaprintf() to simplify code changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering DMI card names ASoC: bytcr-rt5640: register DMI names for card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++ sound/soc/soc-core.c | 67 +++++++++++++++++++++++++++++++++++ 3 files changed, 83 insertions(+) -- 2.5.0

3 7

[alsa-devel] [PATCH] ALSA: intel8x0: Drop superfluous VM checks
by Takashi Iwai 06 Apr '16

06 Apr '16

intel8x0 driver has the inside_vm check for skipping a buggy hardware workaround in the PCM pointer callback in the commit [228cf79376f1: ALSA: intel8x0: Improve performance in virtual environment]. This was originally applied to all devices on known VMs, but the code was switched to use the PCI ID matching for applying to only known devices (KVM and Parallels), in order to avoid applying wrongly to VT-d and other such cases, in the commit [7fb4f392bd27: ALSA: intel8x0: improve virtual environment detection]. Meanwhile, the original VM check was kept even after switching to the PCI ID matching. It was partly because we weren't 100% sure whether we had covered all well, and partly because this would help identifying the issue once when a user of another VM hit the same problem or a regression. Currently the VM check is used only for showing the kernel message that the VM-optimization isn't applied, and the VM check itself doesn't change the actual driver behavior at all. Despite the relatively safe driver behavior, the code caught attention of developers badly and brought many confusion / misunderstanding. Since we've got neither regression nor enhancement report for other VMs for five years long, it's likely safe to drop this superfluous VM check now. The module option is still kept, so if a user still needs to adjust, it can be applied as was. Cc: Andy Lutomirski <luto(a)amacapital.net> Cc: Luis R. Rodriguez <mcgrof(a)kernel.org> Cc: Konstantin Ozerkov <kozerkov(a)parallels.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/intel8x0.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8151318a69a2..9720a30dbfff 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -42,12 +42,6 @@ #include <asm/pgtable.h> #include <asm/cacheflush.h> -#ifdef CONFIG_KVM_GUEST -#include <linux/kvm_para.h> -#else -#define kvm_para_available() (0) -#endif - MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); MODULE_LICENSE("GPL"); @@ -2972,25 +2966,17 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) goto fini; } - /* detect KVM and Parallels virtual environments */ - result = kvm_para_available(); -#ifdef X86_FEATURE_HYPERVISOR - result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); -#endif - if (!result) - goto fini; - /* check for known (emulated) devices */ + result = 0; if (pci->subsystem_vendor == PCI_SUBVENDOR_ID_REDHAT_QUMRANET && pci->subsystem_device == PCI_SUBDEVICE_ID_QEMU) { /* KVM emulated sound, PCI SSID: 1af4:1100 */ msg = "enable KVM"; + result = 1; } else if (pci->subsystem_vendor == 0x1ab8) { /* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */ msg = "enable Parallels VM"; - } else { - msg = "disable (unknown or VT-d) VM"; - result = 0; + result = 1; } fini: -- 2.7.4

3 2

[alsa-devel] [PATCH 0/3 v3] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 05 Apr '16

05 Apr '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) Jose Abreu (3): drm/i2c/adv7511: Rename and move to separate folder drm/i2c/adv7511: Add audio support ASoC: dwc: Unmask I2S interrupts only for enabled channels .../bindings/display/bridge/adi,adv7511.txt | 3 + drivers/gpu/drm/i2c/Kconfig | 6 +- drivers/gpu/drm/i2c/Makefile | 2 +- drivers/gpu/drm/i2c/adv7511/Kconfig | 18 ++ drivers/gpu/drm/i2c/adv7511/Makefile | 3 + drivers/gpu/drm/i2c/{ => adv7511}/adv7511.h | 53 ++++ drivers/gpu/drm/i2c/adv7511/adv7511_audio.c | 310 +++++++++++++++++++++ .../drm/i2c/{adv7511.c => adv7511/adv7511_core.c} | 43 +-- include/sound/soc-dai.h | 1 + sound/soc/dwc/designware_i2s.c | 5 +- 10 files changed, 406 insertions(+), 38 deletions(-) create mode 100644 drivers/gpu/drm/i2c/adv7511/Kconfig create mode 100644 drivers/gpu/drm/i2c/adv7511/Makefile rename drivers/gpu/drm/i2c/{ => adv7511}/adv7511.h (90%) create mode 100644 drivers/gpu/drm/i2c/adv7511/adv7511_audio.c rename drivers/gpu/drm/i2c/{adv7511.c => adv7511/adv7511_core.c} (97%) -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: fsl-asoc-card: add wm8960 support
by Zidan Wang 05 Apr '16

05 Apr '16

add wm8960 support for fsl-asoc-card Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 040362f..5aeb6ed 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" +#include "../codecs/wm8960.h" #define RX 0 #define TX 1 @@ -479,6 +480,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { + codec_dai_name = "wm8960-hifi"; + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; @@ -582,6 +589,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, + { .compatible = "fsl,imx-audio-wm8960", }, {} }; -- 1.9.1

4 4

[alsa-devel] [RFC PATCH v2 0/5] Media Device Allocator API
by Shuah Khan 05 Apr '16

05 Apr '16

There are known problems with media device life time management. When media device is released while an media ioctl is in progress, ioctls fail with use-after-free errors and kernel hangs in some cases. Media Device can be in any the following states: - Allocated - Registered (could be tied to more than one driver) - Unregistered, not in use (media device file is not open) - Unregistered, in use (media device file is not open) - Released When media device belongs to more than one driver, registrations should be tracked to avoid unregistering when one of the drivers does unregister. A new num_drivers field in the struct media_device covers this case. The media device should be unregistered only when the last unregister occurs with num_drivers count zero. When a media device is in use when it is unregistered, it should not be released until the application exits when it detects the unregistered status. Media device that is in use when it is unregistered is moved to to_delete_list. When the last unregister occurs, media device is unregistered and becomes an unregistered, still allocated device. Unregister marks the device to be deleted. When media device belongs to more than one driver, as both drivers could be unbound/bound, driver should not end up getting stale media device that is on its way out. Moving the unregistered media device to to_delete_list helps this case as well. I ran bind/unbind loop tests on uvcvideo, au0828, and snd-usb-audio while running application that does ioctls. Didn't see any use-after-free errors on media device. A couple of known issues seen: 1. When application exits, cdev_put() gets called after media device is released. This is a known issue to resolve and Media Device Allocator can't solve this one. 2. When au0828 module is removed and then ioctls fail when cdev_get() looks for the owning module as au0828 is very often the module that owns the media devnode. This is a cdev related issue that needs to be resolved and Media Device Allocator can't solve this one. Shuah Khan (5): media: Add Media Device Allocator API media: Add driver count to keep track of media device registrations media: uvcvideo change to use Media Device Allocator API media: au0828 change to use Media Device Allocator API sound/usb: Use Media Controller API to share media resources drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 154 ++++++++++++++++ drivers/media/media-device.c | 49 ++++- drivers/media/media-devnode.c | 10 +- drivers/media/usb/au0828/au0828-core.c | 40 +++-- drivers/media/usb/au0828/au0828.h | 1 + drivers/media/usb/uvc/uvc_driver.c | 36 ++-- drivers/media/usb/uvc/uvcvideo.h | 3 +- include/media/media-dev-allocator.h | 116 ++++++++++++ include/media/media-device.h | 31 ++++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 320 +++++++++++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 21 files changed, 862 insertions(+), 37 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.5.0

3 8

[alsa-devel] [RFC PATCH 0/4] Media Device Allocator API
by Shuah Khan 05 Apr '16

05 Apr '16

There are known problems with media device life time management. When media device is released while an media ioctl is in progress, ioctls fail with use-after-free errors and kernel hangs in some cases. Media Device can be in any the following states: - Allocated - Registered (could be tied to more than one driver) - Unregistered, not in use (media device file is not open) - Unregistered, in use (media device file is not open) - Released When media device belongs to more than one driver, registrations should be refcounted to avoid unregistering when one of the drivers does unregister. A refcount field in the struct media_device covers this case. Unregister on a Media Allocator media device is a kref_put() call. The media device should be unregistered only when the last unregister occurs. When a media device is in use when it is unregistered, it should not be released until the application exits when it detects the unregistered status. Media device that is in use when it is unregistered is moved to to_delete_list. When the last unregister occurs, media device is unregistered and becomes an unregistered, still allocated device. Unregister marks the device to be deleted. When media device belongs to more than one driver, as both drivers could be unbound/bound, driver should not end up getting stale media device that is on its way out. Moving the unregistered media device to to_delete_list helps this case as well. I am sending this RFC series out for review. I tested to verify media ioctls don't fail when media device is unregistered and it gets released when the last reference goes. I still need to do more testing. I didb't fully test if the media device gets deleted in all cases. So please consider this as work in progress. I decided to send this out to get early review to see if this solution is viable. Shuah Khan (4): media: Add Media Device Allocator API media: Add Media Device Allocator API documentation media: Add refcount to keep track of media device registrations drivers: change au0828, uvcvideo, snd-usb-audio to use Media Device Allocator drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 153 +++++++++++++++++++++++++++++++++ drivers/media/media-device.c | 53 ++++++++++++ drivers/media/media-devnode.c | 3 + drivers/media/usb/au0828/au0828-core.c | 7 +- drivers/media/usb/au0828/au0828.h | 1 + drivers/media/usb/uvc/uvc_driver.c | 32 ++++--- drivers/media/usb/uvc/uvcvideo.h | 3 +- include/media/media-dev-allocator.h | 113 ++++++++++++++++++++++++ include/media/media-device.h | 32 +++++++ sound/usb/media.c | 10 ++- sound/usb/media.h | 1 + 12 files changed, 390 insertions(+), 21 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h -- 2.5.0

3 15

[alsa-devel] [PATCH 0/3 v2] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 05 Apr '16

05 Apr '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) Jose Abreu (3): drm/i2c/adv7511: Add audio support ASoC: dwc: Add I2S HDMI audio support arc: axs10x: Add support for Designware I2S on DT arch/arc/boot/dts/axs10x_mb.dtsi | 49 +- drivers/gpu/drm/i2c/Kconfig | 11 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 ----------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 +++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 385 ++++++++++++- 10 files changed, 1788 insertions(+), 1041 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c -- 1.9.1

6 21

[alsa-devel] [PATCH v2 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 05 Apr '16

05 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. History: v2: add check on string length, use strtol() to get hex value, and fix memory leak. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 +++- include/topology.h | 79 +++++++- src/topology/data.c | 496 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 695 insertions(+), 8 deletions(-) -- 2.5.0

4 13

Re: [alsa-devel] [PATCH v2- ALSA 2/3] HDA - create headset mic jack which use headphone pin sense
by Justin Monk 05 Apr '16

05 Apr '16

Team, I was wondering when this patch would be available, I figured it would already be in the down stream. If not I would love to test it. I have minimal programming skills but would love to help. Kernel Bug Tracker – Bug 107851 4.2.0-18-generic currently running 4.4.6-300.fc23.x86_64 Thanks, Justin Monk

2 1

[alsa-devel] Sound: BUG: KASAN: use-after-free in kill_fasync
by Baozeng Ding 05 Apr '16

05 Apr '16

Hi all, I've got the following report (use-after-free in kill_fasync) while running syzkaller. Unfortunately no reproducer.The kernel version is 4.5 (on Mar 16 commit 09fd671ccb2475436bd5f597f751ca4a7d177aea). ================================================================== BUG: KASAN: use-after-free in kill_fasync+0x3fb/0x420 at addr ffff880067691d88 Read of size 8 by task swapper/2/0 ============================================================================= BUG kmalloc-2048 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- Disabling lock debugging due to kernel taint INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446678412249576073 cpu=2245704320 pid=-1 [< inline >] kmalloc /kernel/include/linux/slab.h:472 [< inline >] kzalloc /kernel/include/linux/slab.h:616 [< none >] snd_pcm_attach_substream+0x3b4/0xb10 /kernel/sound/core/pcm.c:966 [< none >] ___slab_alloc+0x4c7/0x500 /kernel/mm/slub.c:2446 [< none >] __slab_alloc+0x4c/0x90 /kernel/mm/slub.c:2475 [< inline >] slab_alloc_node /kernel/mm/slub.c:2538 [< inline >] slab_alloc /kernel/mm/slub.c:2580 [< none >] kmem_cache_alloc_trace+0x262/0x300 /kernel/mm/slub.c:2597 [< inline >] kmalloc /kernel/include/linux/slab.h:472 [< inline >] kzalloc /kernel/include/linux/slab.h:616 [< none >] snd_pcm_attach_substream+0x3b4/0xb10 /kernel/sound/core/pcm.c:966 [< none >] snd_pcm_open_substream+0x84/0x450 /kernel/sound/core/pcm_native.c:2262 [< inline >] snd_pcm_oss_open_file /kernel/sound/core/oss/pcm_oss.c:2346 [< none >] snd_pcm_oss_open.part.17+0x5a4/0x1100 /kernel/sound/core/oss/pcm_oss.c:2428 [< none >] snd_pcm_oss_open+0x35/0x50 /kernel/sound/core/oss/pcm_oss.c:2392 [< none >] soundcore_open+0x30f/0x640 /kernel/sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 /kernel/fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 /kernel/fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 /kernel/fs/open.c:853 [< inline >] do_last /kernel/fs/namei.c:3258 [< none >] path_openat+0x4837/0x5830 /kernel/fs/namei.c:3394 [< none >] do_filp_open+0x18e/0x250 /kernel/fs/namei.c:3429 [< none >] do_sys_open+0x201/0x420 /kernel/fs/open.c:1022 [< inline >] SYSC_open /kernel/fs/open.c:1040 [< none >] SyS_open+0x2d/0x40 /kernel/fs/open.c:1035 INFO: Freed in 0x10000b076 age=18446678416544543380 cpu=0 pid=0 [< none >] snd_pcm_detach_substream+0x134/0x280 /kernel/sound/core/pcm.c:1017 [< none >] __slab_free+0x1e8/0x300 /kernel/mm/slub.c:2657 [< inline >] slab_free /kernel/mm/slub.c:2810 [< none >] kfree+0x24e/0x2d0 /kernel/mm/slub.c:3661 [< none >] snd_pcm_detach_substream+0x134/0x280 /kernel/sound/core/pcm.c:1017 [< none >] snd_pcm_release_substream.part.38+0x219/0x2f0 /kernel/sound/core/pcm_native.c:2250 [< none >] snd_pcm_release_substream+0x59/0x70 /kernel/sound/core/pcm_native.c:2251 [< none >] snd_pcm_oss_release_file+0x45/0xb0 /kernel/sound/core/oss/pcm_oss.c:2305 [< none >] snd_pcm_oss_release+0xfa/0x250 /kernel/sound/core/oss/pcm_oss.c:2485 [< none >] __fput+0x236/0x780 /kernel/fs/file_table.c:208 [< none >] ____fput+0x15/0x20 /kernel/fs/file_table.c:244 [< none >] task_work_run+0x16b/0x200 /kernel/kernel/task_work.c:115 [< inline >] exit_task_work /kernel/include/linux/task_work.h:21 [< none >] do_exit+0x87f/0x2c90 /kernel/kernel/exit.c:748 [< none >] do_group_exit+0x108/0x330 /kernel/kernel/exit.c:878 [< inline >] SYSC_exit_group /kernel/kernel/exit.c:889 [< none >] SyS_exit_group+0x1d/0x20 /kernel/kernel/exit.c:887 [< none >] entry_SYSCALL_64_fastpath+0x23/0xc1 /kernel/arch/x86/entry/entry_64.S:207 INFO: Slab 0xffffea00019da400 objects=13 used=8 fp=0xffff880067691be0 flags=0x5fffc0000004080 INFO: Object 0xffff880067691bd8 @offset=7128 fp=0xbbbbbbbbbbbbbbbb Call Trace: <IRQ> [< inline >] __dump_stack /kernel/lib/dump_stack.c:15 <IRQ> [<ffffffff82945051>] dump_stack+0xb3/0x112 /kernel/lib/dump_stack.c:51 [<ffffffff817009ad>] print_trailer+0x10d/0x190 /kernel/mm/slub.c:668 [<ffffffff817074af>] object_err+0x2f/0x40 /kernel/mm/slub.c:675 [< inline >] print_address_description /kernel/mm/kasan/report.c:138 [<ffffffff81709ca5>] kasan_report_error+0x215/0x530 /kernel/mm/kasan/report.c:236 [< inline >] ? spin_lock /kernel/include/linux/spinlock.h:302 [<ffffffff84a1c550>] ? snd_pcm_stream_lock+0x80/0xd0 /kernel/sound/core/pcm_native.c:104 [< inline >] kasan_report /kernel/mm/kasan/report.c:259 [<ffffffff8170a0be>] __asan_report_load8_noabort+0x3e/0x40 /kernel/mm/kasan/report.c:280 [<ffffffff81792a0b>] ? kill_fasync+0x3fb/0x420 /kernel/fs/fcntl.c:729 [<ffffffff81792a0b>] kill_fasync+0x3fb/0x420 /kernel/fs/fcntl.c:729 [<ffffffff84a36e08>] snd_pcm_period_elapsed+0x1c8/0x230 /kernel/sound/core/pcm_lib.c:1890 [<ffffffff84b6e2e0>] ? get_bound_vga.isra.21.part.22+0x140/0x140 /kernel/sound/pci/hda/hda_intel.c:1327 [<ffffffff84b0637d>] stream_update+0xad/0xe0 /kernel/sound/pci/hda/hda_controller.c:923 [<ffffffff84c1583e>] snd_hdac_bus_handle_stream_irq+0x24e/0x350 /kernel/sound/hda/hdac_controller.c:449 [<ffffffff84b062d0>] ? azx_init_chip+0x100/0x100 /kernel/sound/pci/hda/hda_controller.c:890 [<ffffffff84b04d24>] ? azx_interrupt+0x14/0x4d0 /kernel/sound/pci/hda/hda_controller.c:929 [<ffffffff84b04eee>] azx_interrupt+0x1de/0x4d0 /kernel/sound/pci/hda/hda_controller.c:954 [<ffffffff84b04d10>] ? azx_stop_chip+0x20/0x20 /kernel/sound/pci/hda/hda_controller.c:908 [<ffffffff8143d9e3>] handle_irq_event_percpu+0xf3/0x790 /kernel/kernel/irq/handle.c:145 [<ffffffff8143e127>] handle_irq_event+0xa7/0x140 /kernel/kernel/irq/handle.c:192 [<ffffffff81447f81>] handle_edge_irq+0x1e1/0x8d0 /kernel/kernel/irq/chip.c:623 [< inline >] generic_handle_irq_desc /kernel/include/linux/irqdesc.h:146 [<ffffffff811ad739>] handle_irq+0x109/0x2a0 /kernel/arch/x86/kernel/irq_64.c:78 [< inline >] ? rcu_lock_release /kernel/include/linux/rcupdate.h:491 [< inline >] ? rcu_read_unlock /kernel/include/linux/rcupdate.h:926 [< inline >] ? __atomic_notifier_call_chain /kernel/kernel/notifier.c:184 [<ffffffff81369c4f>] ? atomic_notifier_call_chain+0xbf/0x140 /kernel/kernel/notifier.c:193 [<ffffffff81369b90>] ? __atomic_notifier_call_chain+0x150/0x150 /kernel/include/linux/rcupdate.h:922 [<ffffffff811ac36d>] do_IRQ+0x7d/0x1a0 /kernel/arch/x86/kernel/irq.c:240 [<ffffffff85dac40c>] common_interrupt+0x8c/0x8c /kernel/arch/x86/entry/entry_64.S:454 <EOI> [<ffffffff812245c6>] ? native_safe_halt+0x6/0x10 /kernel/./arch/x86/include/asm/irqflags.h:49 [<ffffffff81405ced>] ? trace_hardirqs_on+0xd/0x10 /kernel/kernel/locking/lockdep.c:2635 [< inline >] arch_safe_halt /kernel/./arch/x86/include/asm/paravirt.h:117 [<ffffffff811bf152>] default_idle+0x22/0x2d0 /kernel/arch/x86/kernel/process.c:307 [<ffffffff811c053a>] arch_cpu_idle+0xa/0x10 /kernel/arch/x86/kernel/process.c:298 [<ffffffff813eb618>] default_idle_call+0x48/0x70 /kernel/kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call /kernel/kernel/sched/idle.c:151 [< inline >] cpu_idle_loop /kernel/kernel/sched/idle.c:242 [<ffffffff813ebaa7>] cpu_startup_entry+0x467/0x600 /kernel/kernel/sched/idle.c:291 [<ffffffff81201432>] start_secondary+0x2b2/0x380 /kernel/arch/x86/kernel/smpboot.c:259 [<ffffffff81201180>] ? set_cpu_sibling_map+0x18a0/0x18a0 /kernel/include/linux/topology.h:80 Memory state around the buggy address: ffff880067691c80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff880067691d00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >ffff880067691d80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff880067691e00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff880067691e80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== It seems that in function snd_pcm_detach_substream, when freeing runtime structure, it did not handle its fasync data pointer. Any ideas? Thanks. Best Regards, Baozeng Ding

2 1

[alsa-devel] [PATCH v2 0/2] conf: topology: Add the conf for SKL-rt286
by Vinod Koul 05 Apr '16

05 Apr '16

From: Shreyas NC <shreyas.nc(a)intel.com> The SKL RT286 I2S topology conf is provided here. The first patch adds that along with binary data for modules. The Second patch provides source of those binayy blobs and allows them to be recompiled. Shreyas NC (2): conf: topology: Add Skylake i2s conf conf: topology: Generate Private data binary blobs configure.ac | 1 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 4 + src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/data/Makefile.am | 4 + src/conf/topology/sklrt286/data/README | 2 + src/conf/topology/sklrt286/data/compile_local | 16 + src/conf/topology/sklrt286/data/configure.ac | 30 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_data_local.h | 115 ++ .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 29 files changed, 2420 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin create mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/README create mode 100755 src/conf/topology/sklrt286/data/compile_local create mode 100644 src/conf/topology/sklrt286/data/configure.ac create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data_local.h create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin create mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin create mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin create mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 2.4.11

1 0

[alsa-devel] Getting rid of inside_vm in intel8x0
by Andy Lutomirski 04 Apr '16

04 Apr '16

Would it be possible to revert: commit 228cf79376f13b98f2e1ac10586311312757675c Author: Konstantin Ozerkov <kozerkov(a)parallels.com> Date: Wed Oct 26 19:11:01 2011 +0400 ALSA: intel8x0: Improve performance in virtual environment Presumably one or more of the following is true: a) The inside_vm == true case is just an optimization and should apply unconditionally. b) The inside_vm == true case is incorrect and should be fixed or disabled. c) The inside_vm == true case is a special case that makes sense then IO is very very slow but doesn't make sense when IO is fast. If so, why not literally measure the time that the IO takes and switch over to the "inside VM" path when IO is slow? There are a pile of nonsensical "are we in a VM" checks of various sorts scattered throughout the kernel, they're all a mess to maintain (there are lots of kinds of VMs in the world, and Linux may not even know it's a guest), and, in most cases, it appears that the correct solution is to delete the checks. I just removed a nasty one in the x86_32 entry asm, and this one is written in C so it should be a piece of cake :) --Andy

5 11

[alsa-devel] [PATCH v3] ALSA: hda - Update chmap tlv to report sink's capability
by Subhransu S. Prusty 04 Apr '16

04 Apr '16

The existing TLV callback implementation copies all of the cea_channel_speaker_allocation map table to the TLV container irrespective of what is reported by sink. This is of little use to the userspace application. With this patch, it parses the spk_alloc block as queried from the ELD, and copies only the corresponding mapping channel allocation entries from the cea channel speaker allocation table. Thus the user can parse the TLV container to identify sink's capability and set the channel map accordingly. It shouldn't impact the behavior in AMD chipset, as this makes use of already parsed spk alloc block to calculate the channel map. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- changes in v3: Copy all possible channel map entries w.r.t. sink's capability include/sound/hda_chmap.h | 2 ++ sound/hda/hdmi_chmap.c | 44 ++++++++++++++++++++++++++++++++++++++++---- sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ 3 files changed, 55 insertions(+), 4 deletions(-) diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h index e20d219..babd445 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { int (*chmap_validate)(struct hdac_chmap *hchmap, int ca, int channels, unsigned char *chmap); + int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx); + void (*get_chmap)(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap); void (*set_chmap)(struct hdac_device *hdac, int pcm_idx, diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec862..c6c75e7 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static int spk_mask_from_spk_alloc(int spk_alloc) +{ + int i; + int spk_mask = eld_speaker_allocation_bits[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + return spk_mask; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; int chs, count = 0; + unsigned long max_chs; + int type; + int spk_alloc, spk_mask; if (size < 8) return -ENOMEM; @@ -639,40 +656,59 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + spk_mask = spk_mask_from_spk_alloc(spk_alloc); + + max_chs = hweight_long(spk_mask); + + for (chs = 2; chs <= max_chs; chs++) { int i; struct hdac_cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); unsigned int tlv_chmap[8]; - if (type < 0) + if (cap->channels != chs) + continue; + + if (!(cap->spk_mask == (spk_mask & cap->spk_mask))) continue; + + type = chmap->ops.chmap_cea_alloc_validate_get_type( + chmap, cap, chs); + if (type < 0) + return -ENODEV; if (size < 8) return -ENOMEM; + if (put_user(type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; + dst += 2; size -= 8; count += 8; + if (size < chs_bytes) return -ENOMEM; + size -= chs_bytes; count += chs_bytes; chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) return -EFAULT; dst += chs; } } + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..dd25f2c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1838,6 +1838,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2249,6 +2261,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) spec->chmap.ops.get_chmap = hdmi_get_chmap; spec->chmap.ops.set_chmap = hdmi_set_chmap; spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, codec->spec = spec; hdmi_array_init(spec, 4); -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320
by Takashi Iwai 04 Apr '16

04 Apr '16

Phoenix Audio TMX320 gives the similar error when the sample rate is asked: usb 2-1.3: 2:1: cannot get freq at ep 0x85 usb 2-1.3: 1:1: cannot get freq at ep 0x2 .... Add the corresponding USB-device ID (1de7:0014) to snd_usb_get_sample_rate_quirk() list. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6178bb5d0731..24c7c2311b47 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,6 +1137,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } -- 2.7.4

1 0

Re: [alsa-devel] Getting rid of inside_vm in intel8x0
by Takashi Iwai 04 Apr '16

04 Apr '16

On Mon, 04 Apr 2016 11:05:43 +0200, George Dunlap wrote: > > On 02/04/16 13:57, Andy Lutomirski wrote: > > On Fri, Apr 1, 2016 at 10:33 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> On Sat, 02 Apr 2016 00:28:31 +0200, > >> Luis R. Rodriguez wrote: > >>> If the former, could a we somehow detect an emulated device other than through > >>> this type of check ? Or could we *add* a capability of some sort to detect it > >>> on the driver ? This would not address the removal, but it could mean finding a > >>> way to address emulation issues. > >>> > >>> If its an IO issue -- exactly what is the causing the delays in IO ? > >> > >> Luis, there is no problem about emulation itself. It's rather an > >> optimization to lighten the host side load, as I/O access on a VM is > >> much heavier. > >> > >>>>>> This is satisfied mostly only on VM, and can't > >>>>>> be measured easily unlike the IO read speed. > >>>>> > >>>>> Interesting, note the original patch claimed it was for KVM and > >>>>> Parallels hypervisor only, but since the code uses: > >>>>> > >>>>> +#if defined(__i386__) || defined(__x86_64__) > >>>>> + inside_vm = inside_vm || boot_cpu_has(X86_FEATURE_HYPERVISOR); > >>>>> +#endif > >>>>> > >>>>> This makes it apply also to Xen as well, this makes this hack more > >>>>> broad, but does is it only applicable when an emulated device is > >>>>> used ? What about if a hypervisor is used and PCI passthrough is > >>>>> used ? > >>>> > >>>> A good question. Xen was added there at the time from positive > >>>> results by quick tests, but it might show an issue if it's running on > >>>> a very old chip with PCI passthrough. But I'm not sure whether PCI > >>>> passthrough would work on such old chipsets at all. > >>> > >>> If it did have an issue then that would have to be special cased, that > >>> is the module parameter would not need to be enabled for such type of > >>> systems, and heuristics would be needed. As you note, fortunately this > >>> may not be common though... > >> > >> Actually this *is* module parametered. If set to a boolean value, it > >> can be applied / skipped forcibly. So, if there has been a problem on > >> Xen, this should have been reported. That's why I wrote it's no > >> common case. This comes from the real experience. > >> > >>> but if this type of work around may be > >>> taken as a precedent to enable other types of hacks in other drivers > >>> I'm very fearful of more hacks later needing these considerations as > >>> well. > >>> > >>>>>>> There are a pile of nonsensical "are we in a VM" checks of various > >>>>>>> sorts scattered throughout the kernel, they're all a mess to maintain > >>>>>>> (there are lots of kinds of VMs in the world, and Linux may not even > >>>>>>> know it's a guest), and, in most cases, it appears that the correct > >>>>>>> solution is to delete the checks. I just removed a nasty one in the > >>>>>>> x86_32 entry asm, and this one is written in C so it should be a piece > >>>>>>> of cake :) > >>>>>> > >>>>>> This cake looks sweet, but a worm is hidden behind the cream. > >>>>>> The loop in the code itself is already a kludge for the buggy hardware > >>>>>> where the inconsistent read happens not so often (only at the boundary > >>>>>> and in a racy way). It would be nice if we can have a more reliably > >>>>>> way to know the hardware buggyness, but it's difficult, > >>>>>> unsurprisingly. > >>>>> > >>>>> The concern here is setting precedents for VM cases sprinkled in the kernel. > >>>>> The assumption here is such special cases are really paper'ing over another > >>>>> type of issue, so its best to ultimately try to root cause the issue in > >>>>> a more generalized fashion. > >>>> > >>>> Well, it's rather bare metal that shows the buggy behavior, thus we > >>>> need to paper over it. In that sense, it's other way round; we don't > >>>> tune for VM. The VM check we're discussing is rather for skipping the > >>>> strange workaround. > >>> > >>> What is it exactly about a VM that enables this work around to be skipped? > >>> I don't quite get it yet. > >> > >> VM -- at least the full one with the sound hardware emulation -- > >> doesn't have the hardware bug. So, the check isn't needed. > > > > Here's the issue, though: asking "am I in a VM" is not a good way to > > learn properties of hardware. Just off the top of my head, here are > > some types of VM and what they might imply about hardware: > > > > Intel Kernel Guard: your sound card is passed through from real hardware. > > > > Xen: could go either way. In dom0, it's likely passed through. In > > domU, it could be passed through or emulated, and I believe this is > > the case for all of the Xen variants. > > > > KVM: Probably emulated, but could be passed through. > > I'm not sure exactly why I was CC'd into this thread, but this is an > important point -- even if you're running in a VM, you may actually have > direct un-emulated IO access to a real (buggy) piece of hardware; in > which case it sounds like you still need the work-around. So > boot_cpu_has(X86_FEATURE_HYPERVISOR) is probably not the right check. The VM check is kept there only to show a kernel message; in case where a similar issue is seen on another VM, user may notice more easily by that. The VM check itself doesn't change any kernel behavior any longer. Takashi

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Update chmap tlv to report sink's capability
by Subhransu S. Prusty 04 Apr '16

04 Apr '16

The existing TLV callback implementation copies all of the cea_channel_speaker_allocation map table to the TLV container irrespective of what is reported by sink. This is of little use to the userspace application. With this patch, it parses the spk_alloc block as queried from the ELD, and copies only the corresponding channel map from the cea channel speaker allocation table. Thus the user can parse the TLV container to identify sink's capability and set the channel map accordingly. It shouldn't impact the behavior in AMD chipset, as this makes use of already parsed spk alloc block to calculate the channel map. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- changes in v2: Updated the commit message to make it more understandable. include/sound/hda_chmap.h | 2 ++ sound/hda/hdmi_chmap.c | 90 +++++++++++++++++++++++++++++----------------- sound/pci/hda/patch_hdmi.c | 13 +++++++ 3 files changed, 72 insertions(+), 33 deletions(-) diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h index e20d219..babd445 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { int (*chmap_validate)(struct hdac_chmap *hchmap, int ca, int channels, unsigned char *chmap); + int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx); + void (*get_chmap)(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap); void (*set_chmap)(struct hdac_device *hdac, int pcm_idx, diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec862..9406203 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,40 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static struct hdac_cea_channel_speaker_allocation *get_cap_from_spk_alloc( + int spk_alloc) +{ + int i, spk_mask = 0; + + if (!spk_alloc) + return &channel_allocations[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* Get num channels using spk mask */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if ((spk_mask & channel_allocations[i].spk_mask) == spk_mask) + return &channel_allocations[i]; + } + + return &channel_allocations[0]; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; - int chs, count = 0; + int chs, count = 0, chs_bytes; + int type; + unsigned int tlv_chmap[8]; + int spk_alloc; + struct hdac_cea_channel_speaker_allocation *cap; if (size < 8) return -ENOMEM; @@ -639,40 +666,37 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { - int i; - struct hdac_cea_channel_speaker_allocation *cap; - - cap = channel_allocations; - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { - int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); - unsigned int tlv_chmap[8]; - - if (type < 0) - continue; - if (size < 8) - return -ENOMEM; - if (put_user(type, dst) || - put_user(chs_bytes, dst + 1)) - return -EFAULT; - dst += 2; - size -= 8; - count += 8; - if (size < chs_bytes) - return -ENOMEM; - size -= chs_bytes; - count += chs_bytes; - chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, - tlv_chmap, chs); - if (copy_to_user(dst, tlv_chmap, chs_bytes)) - return -EFAULT; - dst += chs; - } - } + + if (size < 8) + return -ENOMEM; + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + cap = get_cap_from_spk_alloc(spk_alloc); + chs = cap->channels; + + chs_bytes = chs * 4; + type = chmap->ops.chmap_cea_alloc_validate_get_type(chmap, cap, chs); + if (type < 0) + return -ENODEV; + + if (put_user(type, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + + count += chs_bytes; + chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) + return -EFAULT; + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..dd25f2c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1838,6 +1838,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2249,6 +2261,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) spec->chmap.ops.get_chmap = hdmi_get_chmap; spec->chmap.ops.set_chmap = hdmi_set_chmap; spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, codec->spec = spec; hdmi_array_init(spec, 4); -- 1.9.1

2 3

[alsa-devel] [PATCH 0/3] ASoC: hdac_hdmi: Fix codec power state in S3
by Subhransu S. Prusty 02 Apr '16

02 Apr '16

With this series, the codec power is turned OFF when entering suspend if the playback is already in progress. And during resume the power is restored. Codec driver registers the S3 callbacks with prepare/complete to synchronize with Jack notification from display driver. Codec driver takes care of putting the AFG node to proper power state during S3 and controller driver takes care of managing the codec power. Also setting the power state for AFG node is optimized by removing the timeout loop. The driver instead uses a codec_read, which sets the power state and waits for response before the codec power is turned OFF. Subhransu S. Prusty (3): ASoC: hdac_hdmi: Fix to use dev_pm ops instead soc pm ASoC: hdac_hdmi: Fix codec power state in S3 during playback ASoC: Intel: Skylake: Fix to turn OFF codec power when entering S3 sound/soc/codecs/hdac_hdmi.c | 94 +++++++++++++++++++------------------------ sound/soc/intel/skylake/skl.c | 15 ++++++- 2 files changed, 55 insertions(+), 54 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH v2] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 02 Apr '16

02 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 v1: Use codec dai name to determine the capture path. --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 39 ++++++++++++++++++-- 2 files changed, 41 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..383cba9 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: +- mediatek,audio-codec-dai-capture: audio codec dai on capture path + rt5645-aif1: Default setting. Connect rt5650 I2S1 for capture + rt5645-aif2: Connect rt5650 I2S2 for capture + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + mediatek,audio-codec-dai-capture = "rt5645-aif2"; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..fd2ca66 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_name_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_name_capture, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_dai_name_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_name_capture; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +219,17 @@ 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, + "mediatek,audio-codec-dai-capture")) { + device_property_read_string(&pdev->dev, + "mediatek,audio-codec-dai-capture", + &codec_dai_name_capture); + + mt8173_rt5650_codecs[1].dai_name = codec_dai_name_capture; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 02 Apr '16

02 Apr '16

Driver for TI's TAS5720L/M digital audio amplifiers. The driver should be pretty standard except the optional interrupt-based fault reporting. Some background on the fault reporting since that might be a discussion of interest. The code should have that documented rather well but I wanted to bring it up right away. The TAS5720 devices can report conditions such as over-current (short), over temp, and others that can be important to know in a real-world system. However the device lacks a way to mask the generation of fault interrupts except by shutting it down, which can lead to cases where the device generates SAIF-fault host interrupts every 300us! This is clearly a lot of overhead to be thrown at the SoC (and my bench testing has demonstrated that) so after trying different things to tame this behavior I ended up implementing method to disable/enable the interrupts in the driver itself in combination with rate-limiting the processing of those events. The fact of processing interrupts in a threaded fashion (IRQF_ONESHOT) alone was not enough under certain conditions to prevent the SoC from not being able to get to all interrupts even with an empty threaded handler, and it would freeze and eventually the Kernel would disable the interrupt ("no one cared"). The scheme currently implemented seems to work reliably cycling different audio files over hours also forcing error conditions. But if anybody has a "cleaner" idea how to rate-limit IRQs on a very low level (currently done in the IRQ handler function) I'm absolutely open to suggestions. This all may not be an issue at all when using a faster/lower-latency SoC than what I'm using (BeagleBone Black) but the fact that IRQF_ONESHOT alone was not enough to prevent the system to be brought to its knees (again, even with an empty threaded handler) was a bit of a surprise to me. -- Andreas Dannenberg Texas Instruments Inc Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 37 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 734 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 873 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

3 8

[alsa-devel] [PATCH - aconnect 1/1] Fix aconnect printing card and pid on older kernels
by Adam Goode 01 Apr '16

01 Apr '16

alsa-lib is changed to return a distinct negative value if running on an older kernel. -1 is only returned now if the kernel knows for sure that there is no card or pid. On older kernels, alsa-lib will return -ENOSYS. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 85a7770..bbc5bb3 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -177,13 +177,13 @@ static void print_port(snd_seq_t *seq, snd_seq_client_info_t *cinfo, #ifdef HAVE_SEQ_CLIENT_INFO_GET_CARD card = snd_seq_client_info_get_card(cinfo); #endif - if (card != -1) + if (card >= 0) printf(",card=%d", card); #ifdef HAVE_SEQ_CLIENT_INFO_GET_PID pid = snd_seq_client_info_get_pid(cinfo); #endif - if (pid != -1) + if (pid >= 0) printf(",pid=%d", pid); printf("]\n"); } -- 2.8.0.rc3.226.g39d4020

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Enable i915 binding for gen3/4 HDMI/DP
by Takashi Iwai 01 Apr '16

01 Apr '16

This patch fills the holes and now all i915 HDMI/DP codecs are managed over the audio ELD notifier, finally. The old gen3/gen4 chips have usually only a single pin/converter pair, and the digital port mapping is fixed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 11 +++++++++++ sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++------ 2 files changed, 26 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d0da2508823e..c62a9f830b84 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -128,6 +128,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); * Pin Widget 4 - PORT B (port = 1 in i915 driver) * Pin Widget 5 - PORT C (port = 2 in i915 driver) * Pin Widget 6 - PORT D (port = 3 in i915 driver) + * + * on earlier models: + * Pin Widget 3 - PORT B */ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) { @@ -139,6 +142,14 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) case 0x80862882: /* VLV */ base_nid = 3; break; + case 0x80862801: /* Bearlake */ + case 0x80862802: /* Cantiga */ + case 0x80862803: /* Eaglelake */ + case 0x80862880: /* CedarTrail */ + case 0x808629fb: /* Crestline HDMI */ + if (WARN_ON(pin_nid != 0x03)) + return -1; + return 1; /* only port B */ default: base_nid = 4; break; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4833c7bdd1e8..2a3bf1584f3b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2286,6 +2286,15 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) case 0x80862882: /* VLV */ pin_nid = port + 0x03; break; + case 0x80862801: /* Bearlake */ + case 0x80862802: /* Cantiga */ + case 0x80862803: /* Eaglelake */ + case 0x80862880: /* CedarTrail */ + case 0x808629fb: /* Crestline HDMI */ + if (WARN_ON(port != 1)) /* assume only port B */ + return; + pin_nid = 0x03; + break; default: pin_nid = port + 0x04; break; @@ -2424,7 +2433,7 @@ static int patch_i915_byt_hdmi(struct hda_codec *codec) return 0; } -/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */ +/* Intel Gen3/Gen4, IronLake, SandyBridge and IvyBridge; with eld notifier */ static int patch_i915_cpt_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -3627,9 +3636,9 @@ HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), -HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi), @@ -3638,10 +3647,10 @@ HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), -HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_i915_cpt_hdmi), /* special ID for generic HDMI */ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi), {} /* terminator */ -- 2.7.4

2 7

[alsa-devel] [PATCH] a52: Make compatible with ffmpeg 3.0
by Anatol Pomozov 01 Apr '16

01 Apr '16

audioconvert.h has been deprecated a while ago and got removed in ffmpeg 3.0 (commit 2d40968dd3ff17b12f7). Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- a52/pcm_a52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 88529eb..6fad77d 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -39,7 +39,7 @@ #endif #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 34, 0) -#include <libavutil/audioconvert.h> +#include <libavutil/channel_layout.h> #include <libavutil/mem.h> #define USE_AVCODEC_FRAME #endif -- 2.7.1

3 3

[alsa-devel] External Sound Card Prevents Boot, errors
by Alan Taylor 01 Apr '16

01 Apr '16

I have an external sound card--a Topping VX2. When plugged in to USB, the device prevents Ubuntu from booting properly. It hangs until I remove the USB cord from the device, then Ubuntu boots normally. Once booted, the device can be recognized by Ubuntu, but then seems to nuke PulseAudio, leaving no devices in the Sound Settings. The VX2 uses Via Technologies VT 1620A USB audio controller. I suspect that this may be the issue. I've tried this with two separate VX2 devices, with the same results. I see: -110 connection time out -71 protocol error -19 no device Attaching a Pulse Audio log here. I'm running Ubuntu 15.10. also posted: https://bugzilla.kernel.org/show_bug.cgi?id=115661

1 0

[alsa-devel] [PATCH v3 1/5] ASoC: tas571x: added missing register literals
by Petr Kulhavy 01 Apr '16

01 Apr '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to the register 0x25. Added volatile and read-only register tables into tas5711_regmap_config and tas5717_regmap_config. The chip has 256 registers in total. But from address 0x29 on (0x26 to 0x28 are reserved) the register width varies between 20, 12 and 8 bytes, which the register map is unable to represent. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> v3: added volatile and read-only register tables into regmap configs sound/soc/codecs/tas571x.c | 28 ++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 50 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..d255d65 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -241,6 +245,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -260,6 +284,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -314,6 +340,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
by Sven Eckelmann 01 Apr '16

01 Apr '16

The Lenovo Thinkpad T460s requires the alc_fixup_tpt440_dock as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T460s (20F9CT01WW) using a ThinkPad Ultradock on kernel 4.4.6. Signed-off-by: Sven Eckelmann <sven(a)narfation.org> Tested-by: Simon Wunderlich <sw(a)simonwunderlich.de> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 73978c7..5b85019 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4759,6 +4759,8 @@ enum { ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, + ALC292_FIXUP_TPT460_DOCK, + ALC292_FIXUP_TPT460, }; static const struct hda_fixup alc269_fixups[] = { @@ -5401,6 +5403,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC, }, + [ALC292_FIXUP_TPT460_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440_dock, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, + [ALC292_FIXUP_TPT460] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC292_FIXUP_TPT460_DOCK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5554,7 +5568,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -5649,6 +5663,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, + {.id = ALC292_FIXUP_TPT460_DOCK, .name = "tpt460-dock"}, + {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.8.0.rc3

1 0

[alsa-devel] [PATCH] ALSA: timer: Use mod_timer() for rearming the system timer
by Takashi Iwai 01 Apr '16

01 Apr '16

ALSA system timer backend stops the timer via del_timer() without sync and leaves del_timer_sync() at the close instead. This is because of the restriction by the design of ALSA timer: namely, the stop callback may be called from the timer handler, and calling the sync shall lead to a hangup. However, this also triggers a kernel BUG() when the timer is rearmed immediately after stopping without sync: kernel BUG at kernel/time/timer.c:966! Call Trace: <IRQ> [<ffffffff8239c94e>] snd_timer_s_start+0x13e/0x1a0 [<ffffffff8239e1f4>] snd_timer_interrupt+0x504/0xec0 [<ffffffff8122fca0>] ? debug_check_no_locks_freed+0x290/0x290 [<ffffffff8239ec64>] snd_timer_s_function+0xb4/0x120 [<ffffffff81296b72>] call_timer_fn+0x162/0x520 [<ffffffff81296add>] ? call_timer_fn+0xcd/0x520 [<ffffffff8239ebb0>] ? snd_timer_interrupt+0xec0/0xec0 .... It's the place where add_timer() checks the pending timer. It's clear that this may happen after the immediate restart without sync in our cases. So, the workaround here is just to use mod_timer() instead of add_timer(). This looks like a band-aid fix, but it's a right move, as snd_timer_interrupt() takes care of the continuous rearm of timer. Reported-by: Jiri Slaby <jslaby(a)suse.cz> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index ea4d999113ef..6469bedda2f3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1019,8 +1019,8 @@ static int snd_timer_s_start(struct snd_timer * timer) njiff += timer->sticks - priv->correction; priv->correction = 0; } - priv->last_expires = priv->tlist.expires = njiff; - add_timer(&priv->tlist); + priv->last_expires = njiff; + mod_timer(&priv->tlist, njiff); return 0; } -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Update chmap tlv to report sink's capability
by Subhransu S. Prusty 01 Apr '16

01 Apr '16

Currently channel map controls don't reflect the sink's capability to userspace. It reports the complete hdmi spec channel allocation table to the userspace which is of little use to the userspace applications. With this patch, tlv callback is modified to read the speaker allocation block from sink, and a mapping alsa channel map is presented to the userspace. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hda_chmap.h | 2 ++ sound/hda/hdmi_chmap.c | 90 +++++++++++++++++++++++++++++----------------- sound/pci/hda/patch_hdmi.c | 13 +++++++ 3 files changed, 72 insertions(+), 33 deletions(-) diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h index e20d219..babd445 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { int (*chmap_validate)(struct hdac_chmap *hchmap, int ca, int channels, unsigned char *chmap); + int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx); + void (*get_chmap)(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap); void (*set_chmap)(struct hdac_device *hdac, int pcm_idx, diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec862..9406203 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,40 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static struct hdac_cea_channel_speaker_allocation *get_cap_from_spk_alloc( + int spk_alloc) +{ + int i, spk_mask = 0; + + if (!spk_alloc) + return &channel_allocations[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* Get num channels using spk mask */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if ((spk_mask & channel_allocations[i].spk_mask) == spk_mask) + return &channel_allocations[i]; + } + + return &channel_allocations[0]; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; - int chs, count = 0; + int chs, count = 0, chs_bytes; + int type; + unsigned int tlv_chmap[8]; + int spk_alloc; + struct hdac_cea_channel_speaker_allocation *cap; if (size < 8) return -ENOMEM; @@ -639,40 +666,37 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { - int i; - struct hdac_cea_channel_speaker_allocation *cap; - - cap = channel_allocations; - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { - int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); - unsigned int tlv_chmap[8]; - - if (type < 0) - continue; - if (size < 8) - return -ENOMEM; - if (put_user(type, dst) || - put_user(chs_bytes, dst + 1)) - return -EFAULT; - dst += 2; - size -= 8; - count += 8; - if (size < chs_bytes) - return -ENOMEM; - size -= chs_bytes; - count += chs_bytes; - chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, - tlv_chmap, chs); - if (copy_to_user(dst, tlv_chmap, chs_bytes)) - return -EFAULT; - dst += chs; - } - } + + if (size < 8) + return -ENOMEM; + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + cap = get_cap_from_spk_alloc(spk_alloc); + chs = cap->channels; + + chs_bytes = chs * 4; + type = chmap->ops.chmap_cea_alloc_validate_get_type(chmap, cap, chs); + if (type < 0) + return -ENODEV; + + if (put_user(type, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + + count += chs_bytes; + chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) + return -EFAULT; + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..dd25f2c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1838,6 +1838,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2249,6 +2261,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) spec->chmap.ops.get_chmap = hdmi_get_chmap; spec->chmap.ops.set_chmap = hdmi_set_chmap; spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, codec->spec = spec; hdmi_array_init(spec, 4); -- 1.9.1

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[alsa-devel] [PATCH V4 0/2] ASoC: share more product information with user space
by han.lu＠intel.com 01 Apr '16

01 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, so user space utils such as PA and UCM can distinguish different products. 1. Add a core API. 2. Apply the API to bytcr-rt5640 driver. changes on V4: 1. Replace kmalloc() and snprintf() with ksaprintf() to simplify code changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering DMI card names ASoC: bytcr-rt5640: register DMI names to card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 15 ++++++++++ sound/soc/soc-core.c | 56 +++++++++++++++++++++++++++++++++++ 3 files changed, 74 insertions(+) -- 2.5.0

3 4

[alsa-devel] [PATCH] ALSA: hda - fix front mic problem for a HP desktop
by Hui Wang 01 Apr '16

01 Apr '16

The front mic jack (pink color) can't detect any plug or unplug. After applying this fix, both detecting function and recording function work well. BugLink: https://bugs.launchpad.net/bugs/1564712 Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 73978c7..fefe83f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4759,6 +4759,7 @@ enum { ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, + ALC221_FIXUP_HP_FRONT_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -5401,6 +5402,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC, }, + [ALC221_FIXUP_HP_FRONT_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a19020 }, /* Front Mic */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5506,6 +5514,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: qcom: fix static checker warnings.
by Srinivas Kandagatla 31 Mar '16

31 Mar '16

This patchset fixes 2 static checker warnings reported by Dan Carpenter. Thanks, srini Srinivas Kandagatla (2): ASoC: qcom: Fix uninitialized symbol warning. ASoC: qcom: remove IS_ERR_VALUE usage on int. sound/soc/qcom/lpass-platform.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH 1/2] [media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
by Mauro Carvalho Chehab 31 Mar '16

31 Mar '16

Unfortunately, this patch caused several regressions at au0828 and snd-usb-audio, like this one: https://bugzilla.kernel.org/show_bug.cgi?id=115561 It also showed several troubles at the MC core that handles pretty poorly the memory protections and data lifetime management. So, better to revert it and fix the core before reapplying this change. This reverts commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources")' Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- sound/usb/Kconfig | 4 - sound/usb/Makefile | 2 - sound/usb/card.c | 14 --- sound/usb/card.h | 3 - sound/usb/media.c | 318 ----------------------------------------------- sound/usb/media.h | 72 ----------- sound/usb/mixer.h | 3 - sound/usb/pcm.c | 28 +---- sound/usb/quirks-table.h | 1 - sound/usb/stream.c | 2 - sound/usb/usbaudio.h | 6 - 11 files changed, 5 insertions(+), 448 deletions(-) delete mode 100644 sound/usb/media.c delete mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index d14bf411515b..a452ad7cec40 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,7 +15,6 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE - select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -23,9 +22,6 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. -config SND_USB_AUDIO_USE_MEDIA_CONTROLLER - bool - config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 8dca3c407f5a..2d2d122b069f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,8 +15,6 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o -snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o - snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bbba8c7..3fc63583a537 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,7 +66,6 @@ #include "format.h" #include "power.h" #include "stream.h" -#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -612,11 +611,6 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; - if (quirk->media_device) { - /* don't want to fail when media_snd_device_create() fails */ - media_snd_device_create(chip, intf); - } - usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -673,14 +667,6 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } - /* - * Nice to check quirk && quirk->media_device - * need some special handlings. Doesn't look like - * we have access to quirk here - * Acceses mixer_list - */ - media_snd_device_delete(chip); - /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 34a0898e2238..71778ca4b26a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,8 +105,6 @@ struct snd_usb_endpoint { struct list_head list; }; -struct media_ctl; - struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -158,7 +156,6 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ - struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c deleted file mode 100644 index 93a50d01490c..000000000000 --- a/sound/usb/media.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * media.c - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ - -#include <linux/init.h> -#include <linux/list.h> -#include <linux/mutex.h> -#include <linux/slab.h> -#include <linux/usb.h> - -#include <sound/pcm.h> -#include <sound/core.h> - -#include "usbaudio.h" -#include "card.h" -#include "mixer.h" -#include "media.h" - -static int media_snd_enable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->enable_source) - return mctl->media_dev->enable_source(&mctl->media_entity, - &mctl->media_pipe); - return 0; -} - -static void media_snd_disable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->disable_source) - mctl->media_dev->disable_source(&mctl->media_entity); -} - -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream) -{ - struct media_device *mdev; - struct media_ctl *mctl; - struct device *pcm_dev = &pcm->streams[stream].dev; - u32 intf_type; - int ret = 0; - u16 mixer_pad; - struct media_entity *entity; - - mdev = subs->stream->chip->media_dev; - if (!mdev) - return -ENODEV; - - if (subs->media_ctl) - return 0; - - /* allocate media_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; - mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; - mixer_pad = 1; - } else { - intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; - mctl->media_pad.flags = MEDIA_PAD_FL_SINK; - mixer_pad = 2; - } - mctl->media_entity.name = pcm->name; - media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) - goto free_mctl; - - mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, - MAJOR(pcm_dev->devt), - MINOR(pcm_dev->devt)); - if (!mctl->intf_devnode) { - ret = -ENOMEM; - goto unregister_entity; - } - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &mctl->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - ret = -ENOMEM; - goto devnode_remove; - } - - /* create link between mixer and audio */ - media_device_for_each_entity(entity, mdev) { - switch (entity->function) { - case MEDIA_ENT_F_AUDIO_MIXER: - ret = media_create_pad_link(entity, mixer_pad, - &mctl->media_entity, 0, - MEDIA_LNK_FL_ENABLED); - if (ret) - goto remove_intf_link; - break; - } - } - - subs->media_ctl = mctl; - return 0; - -remove_intf_link: - media_remove_intf_link(mctl->intf_link); -devnode_remove: - media_devnode_remove(mctl->intf_devnode); -unregister_entity: - media_device_unregister_entity(&mctl->media_entity); -free_mctl: - kfree(mctl); - return ret; -} - -void media_snd_stream_delete(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl && mctl->media_dev) { - struct media_device *mdev; - - mdev = subs->stream->chip->media_dev; - if (mdev && media_devnode_is_registered(&mdev->devnode)) { - media_devnode_remove(mctl->intf_devnode); - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - subs->media_ctl = NULL; - } -} - -int media_snd_start_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - return media_snd_enable_source(mctl); - return 0; -} - -void media_snd_stop_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - media_snd_disable_source(mctl); -} - -int media_snd_mixer_init(struct snd_usb_audio *chip) -{ - struct device *ctl_dev = &chip->card->ctl_dev; - struct media_intf_devnode *ctl_intf; - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - struct media_mixer_ctl *mctl; - u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; - int ret; - - if (!mdev) - return -ENODEV; - - ctl_intf = chip->ctl_intf_media_devnode; - if (!ctl_intf) { - ctl_intf = media_devnode_create(mdev, intf_type, 0, - MAJOR(ctl_dev->devt), - MINOR(ctl_dev->devt)); - if (!ctl_intf) - return -ENOMEM; - chip->ctl_intf_media_devnode = ctl_intf; - } - - list_for_each_entry(mixer, &chip->mixer_list, list) { - - if (mixer->media_mixer_ctl) - continue; - - /* allocate media_mixer_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; - mctl->media_entity.name = chip->card->mixername; - mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; - mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; - mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; - media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, - mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) { - kfree(mctl); - return ret; - } - - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &ctl_intf->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - kfree(mctl); - return -ENOMEM; - } - mctl->intf_devnode = ctl_intf; - mixer->media_mixer_ctl = mctl; - } - return 0; -} - -static void media_snd_mixer_delete(struct snd_usb_audio *chip) -{ - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - - if (!mdev) - return; - - list_for_each_entry(mixer, &chip->mixer_list, list) { - struct media_mixer_ctl *mctl; - - mctl = mixer->media_mixer_ctl; - if (!mixer->media_mixer_ctl) - continue; - - if (media_devnode_is_registered(&mdev->devnode)) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - mixer->media_mixer_ctl = NULL; - } - if (media_devnode_is_registered(&mdev->devnode)) - media_devnode_remove(chip->ctl_intf_media_devnode); - chip->ctl_intf_media_devnode = NULL; -} - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) -{ - struct media_device *mdev; - struct usb_device *usbdev = interface_to_usbdev(iface); - int ret; - - mdev = media_device_get_devres(&usbdev->dev); - if (!mdev) - return -ENOMEM; - if (!mdev->dev) { - /* register media device */ - mdev->dev = &usbdev->dev; - if (usbdev->product) - strlcpy(mdev->model, usbdev->product, - sizeof(mdev->model)); - if (usbdev->serial) - strlcpy(mdev->serial, usbdev->serial, - sizeof(mdev->serial)); - strcpy(mdev->bus_info, usbdev->devpath); - mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); - media_device_init(mdev); - } - if (!media_devnode_is_registered(&mdev->devnode)) { - ret = media_device_register(mdev); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't register media device. Error: %d\n", - ret); - return ret; - } - } - - /* save media device - avoid lookups */ - chip->media_dev = mdev; - - /* Create media entities for mixer and control dev */ - ret = media_snd_mixer_init(chip); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't create media mixer entities. Error: %d\n", - ret); - - /* clear saved media_dev */ - chip->media_dev = NULL; - - return ret; - } - return 0; -} - -void media_snd_device_delete(struct snd_usb_audio *chip) -{ - struct media_device *mdev = chip->media_dev; - - media_snd_mixer_delete(chip); - - if (mdev) { - if (media_devnode_is_registered(&mdev->devnode)) - media_device_unregister(mdev); - chip->media_dev = NULL; - } -} diff --git a/sound/usb/media.h b/sound/usb/media.h deleted file mode 100644 index 1dcdcdc5f7aa..000000000000 --- a/sound/usb/media.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * media.h - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ -#ifndef __MEDIA_H - -#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER - -#include <media/media-device.h> -#include <media/media-entity.h> -#include <sound/asound.h> - -struct media_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad; - struct media_pipeline media_pipe; -}; - -/* - * One source pad each for SNDRV_PCM_STREAM_CAPTURE and - * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link - * to AUDIO Source -*/ -#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) - -struct media_mixer_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; - struct media_pipeline media_pipe; -}; - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface); -void media_snd_device_delete(struct snd_usb_audio *chip); -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream); -void media_snd_stream_delete(struct snd_usb_substream *subs); -int media_snd_start_pipeline(struct snd_usb_substream *subs); -void media_snd_stop_pipeline(struct snd_usb_substream *subs); -#else -static inline int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) - { return 0; } -static inline void media_snd_device_delete(struct snd_usb_audio *chip) { } -static inline int media_snd_stream_init(struct snd_usb_substream *subs, - struct snd_pcm *pcm, int stream) - { return 0; } -static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { } -static inline int media_snd_start_pipeline(struct snd_usb_substream *subs) - { return 0; } -static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { } -#endif -#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index f3789446ab9c..3417ef347e40 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,8 +3,6 @@ #include <sound/info.h> -struct media_mixer_ctl; - struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -24,7 +22,6 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; - struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0e4e0640c504..44d178ee9177 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,7 +35,6 @@ #include "pcm.h" #include "clock.h" #include "power.h" -#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -718,14 +717,10 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = media_snd_start_pipeline(subs); - if (ret) - return ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - goto err_ret; + return ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -739,27 +734,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - ret = -EINVAL; - goto err_ret; + return -EINVAL; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; - -err_ret: - media_snd_stop_pipeline(subs); - return ret; } /* @@ -771,7 +761,6 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; - media_snd_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1232,7 +1221,6 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; - int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1246,12 +1234,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - ret = setup_hw_info(runtime, subs); - if (ret == 0) - ret = media_snd_stream_init(subs, as->pcm, direction); - if (ret) - snd_usb_autosuspend(subs->stream->chip); - return ret; + return setup_hw_info(runtime, subs); } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1260,7 +1243,6 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); - media_snd_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9d087b19c70c..c60a776e815d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,7 +2886,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ - .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 51258a15f653..c4dc577ab1bd 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,7 +36,6 @@ #include "format.h" #include "clock.h" #include "stream.h" -#include "media.h" /* * free a substream @@ -53,7 +52,6 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); - media_snd_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a161c7c1b126..b665d85555cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,9 +30,6 @@ * */ -struct media_device; -struct media_intf_devnode; - struct snd_usb_audio { int index; struct usb_device *dev; @@ -63,8 +60,6 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ - struct media_device *media_dev; - struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -115,7 +110,6 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; - bool media_device; const void *data; }; -- 2.5.5

1 0

[alsa-devel] [PATCH] ASoC: qcom: Fix some static checker warnings
by Srinivas Kandagatla 31 Mar '16

31 Mar '16

This patch fixes below static checker warning plus suggestions from Dan. sound/soc/qcom/lpass-platform.c:555 lpass_platform_pcm_new() error: uninitialized symbol 'ret'. sound/soc/qcom/lpass-platform.c 515 csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 516 if (csubstream) { 517 if (v->alloc_dma_channel) 518 data->wrdma_ch = v->alloc_dma_channel(drvdata, 519 SNDRV_PCM_STREAM_CAPTURE); 520 521 if (IS_ERR_VALUE(data->wrdma_ch)) 522 goto capture_alloc_err; wrdma_ch is an int so this should just be: if (data->wrdma_ch < 0) { ret = data->wrdma_ch; goto capture_alloc_err; } Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-platform.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 6e86654..db000c6 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -474,7 +474,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) struct lpass_data *drvdata = snd_soc_platform_get_drvdata(soc_runtime->platform); struct lpass_variant *v = drvdata->variant; - int ret; + int ret = -EINVAL; struct lpass_pcm_data *data; size_t size = lpass_platform_pcm_hardware.buffer_bytes_max; @@ -491,7 +491,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->rdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_PLAYBACK); - if (IS_ERR_VALUE(data->rdma_ch)) + if (data->rdma_ch < 0) return data->rdma_ch; drvdata->substream[data->rdma_ch] = psubstream; @@ -518,8 +518,10 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->wrdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_CAPTURE); - if (IS_ERR_VALUE(data->wrdma_ch)) + if (data->wrdma_ch < 0) { + ret = data->wrdma_ch; goto capture_alloc_err; + } drvdata->substream[data->wrdma_ch] = csubstream; -- 2.5.0

2 2

[alsa-devel] [PATCH v2 1/5] Sound: SOC: TAS571x: added missing register literals
by Petr Kulhavy 31 Mar '16

31 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to register 0x25 Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> sound/soc/codecs/tas571x.c | 4 ++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..346d3da 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

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[alsa-devel] [PATCH v4 1/3] ASoC: tas571x: added missing register literals
by Petr Kulhavy 31 Mar '16

31 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to the register 0x25. Added volatile and read-only register tables into tas5711_regmap_config and tas5717_regmap_config. The chip has 256 registers in total. But from address 0x29 on (0x26 to 0x28 are reserved) the register width varies between 20, 12 and 8 bytes, which the register map cannot represent. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> v3: added volatile and read-only register tables into regmap configs v4: no change, the set v3 resubmitted without the already accepted patches sound/soc/codecs/tas571x.c | 28 ++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 50 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index aafee9bbe01a..ef6c8d9b251a 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -259,6 +263,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -278,6 +302,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -332,6 +358,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471232cd..cf800c364f0f 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8825: assign DAC Ch to match headset L/R
by John Hsu 31 Mar '16

31 Mar '16

The default value of DAC channel select is reverse in codec. For normal usage, switch the channel select when codec bootup. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 5 +++++ sound/soc/codecs/nau8825.h | 6 ++++++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..1a99a25 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -905,6 +905,11 @@ static void nau8825_init_regs(struct nau8825 *nau8825) NAU8825_RDAC_CLK_DELAY_MASK | NAU8825_RDAC_VREF_MASK, (0x2 << NAU8825_RDAC_CLK_DELAY_SFT) | (0x3 << NAU8825_RDAC_VREF_SFT)); + /* Config L/R channel */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_R); } static const struct regmap_config nau8825_regmap_config = { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 5fe009d..fcf0d0a 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -256,9 +256,15 @@ /* DACL_CTRL (0x33) */ #define NAU8825_DACL_CH_SEL_SFT 9 +#define NAU8825_DACL_CH_SEL_MASK (0x1 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_L (0x0 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_R (0x1 << NAU8825_DACL_CH_SEL_SFT) /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +#define NAU8825_DACR_CH_SEL_MASK (0x1 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_L (0x0 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_R (0x1 << NAU8825_DACR_CH_SEL_SFT) /* CLASSG_CTRL (0x50) */ #define NAU8825_CLASSG_TIMER_SFT 8 -- 2.6.4

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix double-free in snd_usb_add_audio_stream()
by Vladis Dronov 31 Mar '16

31 Mar '16

Hello, Takashi, Jaroslav, all, Please, see the research and the following patch on a double-free bug in [snd-usb-audio]. 1) The upstream commits 0f886ca1, 902eb7fd and 447d6275f (many thanks to Takashi Iwai) revealed that there is a double-free bug in [snd-usb-audio] module due to alloc/free logic flaw in snd_usb_add_audio_stream() function. A double-free leads to kernel structure/list/slab corruptions and shortly to null-deref, GPF or lockup. 2.1) Let me describe what happens with the current code in usb_audio_probe(), create_fixed_stream_quirk() and snd_usb_add_audio_stream(): > [ sound/usb/card.c ] > static int usb_audio_probe(struct usb_interface *intf, > const struct usb_device_id *usb_id) > { > ... > if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { > /* need some special handlings */ > err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk); > if (err < 0) > goto __error; > } > ... > __error: > if (chip) { > if (!chip->num_interfaces) > snd_card_free(chip->card); Somewhere in the middle of usb_audio_probe() the function snd_usb_create_quirk() is called, and if it returns with an error and no interfaces were created, the sound card is destroyed with "snd_card_free(chip->card)". 2.2) create_fixed_stream_quirk() is called from snd_usb_create_quirk(): > [ sound/usb/quirks.c ] > static int create_fixed_stream_quirk(struct snd_usb_audio *chip, > struct usb_interface *iface, > struct usb_driver *driver, > const struct snd_usb_audio_quirk *quirk) > { > struct audioformat *fp; > struct usb_host_interface *alts; > struct usb_interface_descriptor *altsd; > ... > fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); > ... > (*) stream = (fp->endpoint & USB_DIR_IN) > (*) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; > (*) err = snd_usb_add_audio_stream(chip, stream, fp); > (*) if (err < 0) > (*) goto error; > > if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber || > fp->altset_idx >= iface->num_altsetting) { > err = -EINVAL; > goto error; > } > alts = &iface->altsetting[fp->altset_idx]; > altsd = get_iface_desc(alts); > if (altsd->bNumEndpoints < 1) { > err = -EINVAL; > goto error; > } > ... > error: > kfree(fp); > kfree(rate_table); > return err; > } 2.3) *fp is allocated and passed to snd_usb_add_audio_stream() where snd_usb_init_substream() is called: > [ sound/usb/stream.c ] > int snd_usb_add_audio_stream(struct snd_usb_audio *chip, > int stream, > struct audioformat *fp) > { > struct snd_usb_stream *as; > struct snd_usb_substream *subs; > struct snd_pcm *pcm; > ... > /* create a new pcm */ > as = kzalloc(sizeof(*as), GFP_KERNEL); > ... > snd_usb_init_substream(as, stream, fp); 2.4) In turn snd_usb_init_substream() adds audioformat *fp by its &fp->list to substream's fmt_list list: > [ sound/usb/stream.c ] > static void snd_usb_init_substream(struct snd_usb_stream *as, > int stream, > struct audioformat *fp) > { > struct snd_usb_substream *subs = &as->substream[stream]; > > INIT_LIST_HEAD(&subs->fmt_list); > ... > list_add_tail(&fp->list, &subs->fmt_list); > subs->num_formats++; 2.5) Things go bad from this point in case snd_usb_add_audio_stream() or the caller go the error path. The bug is that the caller frees (see "error: kfree(fp);" code) *fp on the error path, BUT the pointer to the already-freed memory remains in the substream's fmt_list list. The double-free happens after create_fixed_stream_quirk() returns with an error and usb_audio_probe() calls snd_card_free()->...->snd_usb_audio_pcm_free()->free_substream(). As subs->fmt_list is already corrupted, iterating it with list_for_each_entry_safe() leads to any and unpredictable results. > static void free_substream(struct snd_usb_substream *subs) > { > struct audioformat *fp, *n; > ... > list_for_each_entry_safe(fp, n, &subs->fmt_list, list) { > ... > kfree(fp); 3.1) The crash is reproduceable by faking the USB device with no endpoints as described in https://bugzilla.redhat.com/show_bug.cgi?id=1283355 and https://bugzilla.redhat.com/show_bug.cgi?id=1283358. Please see as a proof the following kernel log with debug printing added to the code. First, *fp is added to fmt_list in snd_usb_init_substream(): [ 322.797223] usb 1-1: new full-speed USB device number 2 using xhci_hcd [ 322.974083] usb 1-1: config 1 interface 0 altsetting 0 has 3 endpoint descriptors, different from the interface descriptor's value: 0 [ 322.987031] usb 1-1: New USB device found, idVendor=045e, idProduct=0283 [ 322.998318] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 323.083913] media: Linux media interface: v0.10 [ 323.231249] init_substream: add_tail() fp=ffff88003364ba80 fp->list.next=ffff8800b1e898b8 fp->list.prev=ffff8800b1e898b8 fmt_list=ffff8800b1e898b8 fmt_list.next=ffff88003364ba80 prev=ffff88003364ba80 num_formats=1 As you can see we have a correct list here with head at fmt_list=ffff8800b1e898b8 and single element fp=ffff88003364ba80. 3.2) The code finds out that there are too few endpoints present and goes the error path (to the "error:" label): [ 323.307927] usb 1-1: too few endpoints [ 323.312964] trace-before-free: substream-0 ffff8800b1e89818 numf 1 fmt_list ffff8800b1e898b8 next ffff88003364ba80 [ 323.353759] fp=ffff88003364ba80 next=ffff8800b1e898b8 prev=ffff8800b1e898b8 rate=(null) chmap=(null) [ 323.362118] struct sane As you can see the substream's fmt_list is sane at this point. 3.3) After "kfree(fp)" in the error path of create_fixed_stream_quirk() *fp is freed, BUT the pointer to the freed memory remains in fmt_list. After *fp is freed the list is corrupted and contains trash: [ 323.371752] KFREE(fp) ffff88003364ba80 [ 323.380383] trace-after-free: substream-0 ffff8800b1e89818 numf 1 fmt_list ffff8800b1e898b8 next ffff88003364ba80 [ 323.400003] fp=ffff88003364ba80 next=ffff88003364bf80 prev=ffff8800b1e898b8 rate=(null) chmap=(null) [ 323.406786] fp=ffff88003364bf80 next= (null) prev= (null) rate=(null) chmap=(null) [ 323.422211] next == NULL: FAIL, struct INSANE [ 323.436706] KFREE(rate_table) (null) 3.4) After error return from create_fixed_stream_quirk() the sound card is destroyed with "snd_card_free(chip->card)" in usb_audio_probe(). In the end free_substream() is called: [ 323.511256] usb 1-1: snd_usb_create_quirk() failed: -22 [ 323.565108] list_for_each_entry_safe(): fp=ffff88003364ba80 n=ffff88003364bf80 [ 323.586337] kfree fp ffff88003364ba80 <<< DOUBLE-FREE [ 323.588509] loop-end: fp=ffff88003364ba80 n=ffff88003364bf80 [ 323.599969] list_for_each_entry_safe(): fp=ffff88003364bf80 n=(null) [ 323.610460] kfree fp ffff88003364bf80 <<< FREEING SOMEONE ELSE'S MEMORY [ 323.613181] loop-end: fp=ffff88003364bf80 n=(null) <<< NULL-PTR DEREF ... [ 324.247113] BUG: unable to handle kernel NULL pointer dereference at (null) [ 324.247533] IP: [<ffffffffc02d40ef>] free_substream.part.0+0xef/0x100 [snd_usb_audio] [ 324.248088] PGD 0 [ 324.248088] Oops: 0000 [#1] SMP [ 324.248088] Modules linked in: snd_usb_audio(+) snd_usbmidi_lib snd_hwdep ... [ 324.248088] CPU: 2 PID: 767 Comm: systemd-udevd Not tainted 4.5.0-vladis #25 [ 324.248088] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 [ 324.248088] task: ffff880034718000 ti: ffff8800b2fbc000 task.ti: ffff8800b2fbc000 [ 324.248088] RIP: 0010:[<ffffffffc02d40ef>] [<ffffffffc02d40ef>] free_substream.part.0+0xef/0x100 [snd_usb_audio] [ 324.248088] RSP: 0018:ffff8800b2fbf898 EFLAGS: 00010217 ... [ 324.248088] Call Trace: [ 324.248088] [<ffffffffc02d43fa>] snd_usb_audio_pcm_free+0x9a/0xa0 [snd_usb_audio] [ 324.248088] [<ffffffffc028d982>] snd_pcm_free+0x32/0xa0 [snd_pcm] [ 324.248088] [<ffffffffc028da02>] snd_pcm_dev_free+0x12/0x20 [snd_pcm] [ 324.248088] [<ffffffffc027d279>] __snd_device_free+0x29/0x70 [snd] [ 324.248088] [<ffffffffc027d660>] snd_device_free_all+0x30/0x60 [snd] [ 324.248088] [<ffffffffc02777a4>] release_card_device+0x34/0x90 [snd] [ 324.248088] [<ffffffff815ae2b2>] device_release+0x32/0x90 [ 324.248088] [<ffffffff81455f8a>] kobject_release+0x7a/0x190 [ 324.248088] [<ffffffff81455e47>] kobject_put+0x27/0x50 [ 324.248088] [<ffffffff815ae5f7>] put_device+0x17/0x20 [ 324.248088] [<ffffffffc0277b57>] snd_card_free+0x67/0x90 [snd] [ 324.248088] [<ffffffffc02c4f14>] usb_audio_probe+0x754/0x9d0 [snd_usb_audio] ... 4.1) The suggested patch consists of 2 changes. First, I suppose we should move the code in create_fixed_stream_quirk() marked as "(*)" (see above) after "if (altsd->bNumEndpoints < 1)" check. This way no allocations is done if we go an error path. I've checked that fp->iface and fp->altset_idx are not changed in snd_usb_add_audio_stream() and functions called from it, so it is safe to swap these 2 pieces of code. 4.2) The problem with double-free still remains. I've verified that all the callers of snd_usb_add_audio_stream() free *fp in case of error: $ git grep snd_usb_add_audio_stream sound/usb/quirks.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) > * goto error; > * error: > * kfree(fp); > * kfree(rate_table); > * return err; sound/usb/quirks.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) { > * kfree(fp); > * return err; > * } sound/usb/stream.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) { > * kfree(fp->rate_table); > * kfree(fp->chmap); > * kfree(fp); > * return err; > * } This means that snd_usb_add_audio_stream() should remove *fp from the substream's fmt_list list on the error path, if it was already added. Such places are: > int snd_usb_add_audio_stream(struct snd_usb_audio *chip, int stream, struct audioformat *fp) > { > struct snd_usb_stream *as; > struct snd_usb_substream *subs; > struct snd_pcm *pcm; > int err; > > list_for_each_entry(as, &chip->pcm_list, list) { > subs = &as->substream[stream]; > ... > list_add_tail(&fp->list, &subs->fmt_list); <<< ADDING fp HERE > subs->num_formats++; > return 0; <<< NO ERROR PATH HERE > } > } > > /* look for an empty stream */ > list_for_each_entry(as, &chip->pcm_list, list) { > subs = &as->substream[stream]; > ... > snd_usb_init_substream(as, stream, fp); <<< ADDING fp HERE, IF add_chmap() FAILS > return add_chmap(as->pcm, stream, subs); <<< fp SHOULD BE REMOVED FROM fmt_list > } > > /* create a new pcm */ > as = kzalloc(sizeof(*as), GFP_KERNEL); > err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, > ... > if (err < 0) { <<< fp IS NOT ADDED YET HERE, SO FINE > kfree(as); > return err; > } > ... > snd_usb_init_substream(as, stream, fp); <<< ADDING fp HERE > ... <<< IF add_chmap() FAILS fp SHOULD > return add_chmap(pcm, stream, &as->substream[stream]); <<< BE REMOVED FROM fmt_list > } add_chmap() itself does not add anything to fmt_list list, so we indeed need to remove only the single list element from the list. Having all the above in mind, the patch follows. 4.3) How to handle possible error paths after successful call to snd_usb_add_audio_stream() in create_fixed_stream_quirk() is d iscussable. Properly it should be like the below, but I believe it is overcomplication here would and stick to a simple error_after_add_audio_stream: label: > error2: > snd_usb_del_audio_stream(...something...); > error: > kfree(fp); > kfree(rate_table); > return err; Best regards, Vladis Dronov | Red Hat, Inc. | Product Security Engineer

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[alsa-devel] [PATCH V3 0/2] ASoC: share more product information with user space
by han.lu＠intel.com 31 Mar '16

31 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, so user space utils such as PA and UCM can distinguish different products. 1. Add a core API. 2. Apply the API to bytcr-rt5640 driver. changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering DMI card names ASoC: bytcr-rt5640: register DMI names to card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 15 ++++++++++ sound/soc/soc-core.c | 53 +++++++++++++++++++++++++++++++++++ 3 files changed, 71 insertions(+) -- 2.5.0

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[alsa-devel] [PATCH v2] ALSA: hda - Bind with i915 only when Intel graphics is present
by Takashi Iwai 31 Mar '16

31 Mar '16

On Skylake and onwards, the HD-audio controller driver needs to bind with i915 for having the control of power well audio domain before actually probing the codec. This leads to the load of i915 driver from the audio driver side. But, there are systems that have no Intel graphics but Nvidia or AMD GPU, although they still use HD-audio bus for the onboard audio codecs. On these, loading the i915 driver is nothing but a useless memory and CPU consumption. A simple way to avoid it is just to look for the Intel graphics PCI entry beforehand, and try to bind with i915 only when such an entry is found. Currently, it assumes the PCI display class. If another class appears, this needs to be extended (although it's very unlikely). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Check also PCI vendor id to match only with Intel Now get out of a series, as this should be applied to 4.6, too. sound/hda/hdac_i915.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d0da2508823e..6800e0c5a38f 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -287,6 +287,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); +/* check whether intel graphics is present */ +static bool i915_gfx_present(void) +{ + static struct pci_device_id ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), + .class = PCI_BASE_CLASS_DISPLAY << 16, + .class_mask = 0xff << 16 }, + {} + }; + return pci_dev_present(ids); +} + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -309,6 +321,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (WARN_ON(hdac_acomp)) return -EBUSY; + if (!i915_gfx_present()) + return -ENODEV; + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); if (!acomp) return -ENOMEM; -- 2.7.4

1 0

[alsa-devel] ALSA license
by Li, Steve 31 Mar '16

31 Mar '16

I am trying to use ALSA in a commercial application. I see that in the introduction page, ALSA is licensed under GPL and LGPL. Can you tell me which part of ALSA is licensed under LGPL? What is the best way to find out this information? Thanks Steve ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

3 3

[alsa-devel] [PATCH] ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
by Maruthi Srinivas Bayyavarapu 31 Mar '16

31 Mar '16

This commit fixes garbled audio on Polaris-10/11 variants Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Acked-by: Christian König <christian.koenig(a)amd.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2624cfe..b680be0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2361,6 +2361,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaae8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaaf0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dpcm: prevent error for paths including static links
by Piotr Stankiewicz 31 Mar '16

31 Mar '16

According to the DPCM documentation using static codec <-> codec links along dynamic ones is a valid use case, however this causes errors of the following form to be printed to the kernel log: ASoC: can't get [playback|caputure] BE for <widget name> ASoC: no BE found for <widget name> This happens when setting up, e.g. a route starting with a dynamic DAI, which passes through a static DAI (say, "Some FE" -> "Some BE" -> "Codec to Modem link"). All DAPM widgets involved in that path will be passed to dpcm_add_paths in order to establish FE <-> BE connections, which will result in dpcm_get_be trying to locate back-ends for dynamic, as well as static links, causing a spurious error to be logged for the latter. This patch changes dpcm_get_be to look up a non front-end runtime for a stream widget, and renames it accordingly. Like this both dynamic and static links can be handled properly in dpcm_add_paths, i.e. an actual missing BE can be distinguished from a normal link. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1af4f23..ff5ea7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1220,47 +1220,47 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) } } -/* get BE for DAI widget and stream */ -static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, +/* get runtime for DAI widget and stream */ +static struct snd_soc_pcm_runtime *dpcm_get_rtd(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { - struct snd_soc_pcm_runtime *be; + struct snd_soc_pcm_runtime *rtd; int i; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - list_for_each_entry(be, &card->rtd_list, list) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (!be->dai_link->no_pcm) + if (rtd->dai_link->dynamic) continue; - if (be->cpu_dai->playback_widget == widget) - return be; + if (rtd->cpu_dai->playback_widget == widget) + return rtd; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; if (dai->playback_widget == widget) - return be; + return rtd; } } } else { - list_for_each_entry(be, &card->rtd_list, list) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (!be->dai_link->no_pcm) + if (rtd->dai_link->dynamic) continue; - if (be->cpu_dai->capture_widget == widget) - return be; + if (rtd->cpu_dai->capture_widget == widget) + return rtd; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; if (dai->capture_widget == widget) - return be; + return rtd; } } } - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", + dev_err(card->dev, "ASoC: can't get %s runtime for %s\n", stream ? "capture" : "playback", widget->name); return NULL; } @@ -1367,15 +1367,15 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; } - /* is there a valid BE rtd for this widget */ - be = dpcm_get_be(card, list->widgets[i], stream); + /* is there a valid rtd for this widget */ + be = dpcm_get_rtd(card, list->widgets[i], stream); if (!be) { - dev_err(fe->dev, "ASoC: no BE found for %s\n", + dev_err(fe->dev, "ASoC: no runtime found for %s\n", list->widgets[i]->name); continue; } - /* make sure BE is a real BE */ + /* check if the runtime is a BE */ if (!be->dai_link->no_pcm) continue; -- 1.7.10.4

3 6

[alsa-devel] [PATCH] ASoC: nau8825: Fix jack detection across suspend
by Ben Zhang 31 Mar '16

31 Mar '16

Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power. Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 126 +++++++++++++++++++++++++-------------------- 1 file changed, 71 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..cb08a35 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -343,9 +343,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -607,6 +610,16 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap) static void nau8825_restart_jack_detection(struct regmap *regmap) { + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + /* this will restart the entire jack detection process including MIC/GND * switching and create interrupts. We have to go from 0 to 1 and back * to 0 to restart. @@ -728,7 +741,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to read irq status\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -1198,33 +1214,74 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return ret; } } - - ret = regcache_sync(nau8825->regmap); - if (ret) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } } - break; case SND_SOC_BIAS_OFF: if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); - - regcache_mark_dirty(nau8825->regmap); break; } return 0; } +#ifdef CONFIG_PM +static int nau8825_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + regcache_cache_only(nau8825->regmap, true); + regcache_mark_dirty(nau8825->regmap); + + return 0; +} + +static int nau8825_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + /* The chip may lose power and reset in S3. regcache_sync restores + * register values including configurations for sysclk, irq, and + * jack/button detection. + */ + regcache_cache_only(nau8825->regmap, false); + regcache_sync(nau8825->regmap); + + /* Check the jack plug status directly. If the headset is unplugged + * during S3 when the chip has no power, there will be no jack + * detection irq even after the nau8825_restart_jack_detection below, + * because the chip just thinks no headset has ever been plugged in. + */ + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + + enable_irq(nau8825->irq); + + /* Run jack detection to check the type (OMTP or CTIA) of the headset + * if there is one. This handles the case where a different type of + * headset is plugged in during S3. This triggers an IRQ iff a headset + * is already plugged in. + */ + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} +#else +#define nau8825_suspend NULL +#define nau8825_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_suspend, + .resume = nau8825_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1334,16 +1391,6 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); - ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "nau8825", nau8825); @@ -1411,36 +1458,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1467,7 +1484,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, -- 2.8.0.rc3.226.g39d4020

3 2

[alsa-devel] [PATCH 0/8] ASoC: Intel: Skylake: Add Broxton DSP support
by Vinod Koul 31 Mar '16

31 Mar '16

Broxton DSP is similar to Skylake with similar DSP capabilities but differs in a few subtle ways. It doesn't have Code Loader DMA and use HDA DMA. The DSP power up/down routines are also bit different in the two platforms. Also the FW needs to be downloaded only once and not on every power sequence as it is cached by DSP in protected memory region. All these changes for new DSP ops are abstracted in a new BXTN-dsp ops file. This patch series starts with moving code around to prepare for new code and also fix whitespace issue in existing code. Since we have different loader ops we move Skylake ones to newly defined ops. Then we add device IDs and lastly the broxton ops. Jayachandran B (1): ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core Jeeja KP (4): ASoC: Intel: add dmabuffer to common sst_dsp ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops ASoC: hdac_hdmi: Add broxton device ID ASoC: Intel: Bxtn: Add Broxton DSP support Senthilnathan Veppur (1): ASoC: Intel: Bxtn: Add Broxton PCI ID Vinod Koul (2): ASoC: Intel: Skylake: Move module id defines ASoC: Intel: Skylake: Fix whitepsace issues sound/soc/codecs/hdac_hdmi.c | 1 + sound/soc/intel/Kconfig | 1 + sound/soc/intel/common/sst-dsp-priv.h | 1 + sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/bxt-sst.c | 328 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 182 +++++++++++++++++- sound/soc/intel/skylake/skl-sst-dsp.c | 14 +- sound/soc/intel/skylake/skl-sst-dsp.h | 20 +- sound/soc/intel/skylake/skl-sst.c | 3 - sound/soc/intel/skylake/skl.c | 7 + sound/soc/intel/skylake/skl.h | 12 +- 11 files changed, 549 insertions(+), 22 deletions(-) create mode 100644 sound/soc/intel/skylake/bxt-sst.c -- 1.9.1

2 16

[alsa-devel] [PATCH] ASoC: intel: add function stub when ACPI is not enabled
by Randy Dunlap 31 Mar '16

31 Mar '16

From: Randy Dunlap <rdunlap(a)infradead.org> Add function stub for "sst_acpi_find_name_from_hid()" when CONFIG_ACPI is not enabled so that the driver will build successfully. This fixes the following build errors: (loadable module) ERROR: "sst_acpi_find_name_from_hid" [sound/soc/intel/boards/snd-soc-sst-bytcr-rt5640.ko] undefined! (or built-in) bytcr_rt5640.c:(.text+0x26fc52): undefined reference to `sst_acpi_find_name_from_hid' Reported-by: Borislav Petkov <bp(a)alien8.de> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/common/sst-acpi.h | 9 +++++++++ 1 file changed, 9 insertions(+) --- linux-next-20160308.orig/sound/soc/intel/common/sst-acpi.h +++ linux-next-20160308/sound/soc/intel/common/sst-acpi.h @@ -12,10 +12,19 @@ * */ +#include <linux/kconfig.h> +#include <linux/stddef.h> #include <linux/acpi.h> /* translation fron HID to I2C name, needed for DAI codec_name */ +#if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +#else +inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +#endif /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines);

2 1

[alsa-devel] [PATCH] ASoC: wm8960: Add I2C dependency
by Fabio Estevam 30 Mar '16

30 Mar '16

Building wm8960 codec without I2C support causes build issues: sound/soc/codecs/wm8960.c:1388:1: warning: data definition has no type or storage class module_i2c_driver(wm8960_i2c_driver); ^ sound/soc/codecs/wm8960.c:1388:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/wm8960.c:1388:1: warning: parameter names (without types) in function declaration sound/soc/codecs/wm8960.c:1378:26: warning: 'wm8960_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver wm8960_i2c_driver = { , so add the I2C dependency to fix this problem. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ba835b6..09e8c39 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -911,6 +911,7 @@ config SND_SOC_WM8955 config SND_SOC_WM8960 tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate -- 1.9.1

2 1

[alsa-devel] [RFC][PATCH] ALSA: firewire-lib: support Linux tracing to dump a part of packet data
by Takashi Sakamoto 30 Mar '16

30 Mar '16

When audio and music units have some quirks in their sequence of packet, it's really hard for non-owners to identify the quirks. To handle the quirks, developers need a dump for sequence of packets at least, however it's difficult to users who have no knowledge and equipment for this purpose. This commit adds tracepoints for this situation. When users encounter the issue, they can dump a part of packet data via Linux tracing framework as long as using drivers in ALSA firewire stack. This commit adds 'snd-firewire-lib' subsystem with 'in_packet' and 'out_packet' events. In the events, the content of CIP headers, the number of included quadlets and the index of packet managed by this module are recorded per packet. This is a sample: $ trace-cmd record -e snd_firewire_lib:out_packet -e snd_firewire_lib:in_packet /sys/kernel/tracing/events/snd_firewire_lib/out_packet/filter /sys/kernel/tracing/events/snd_firewire_lib/in_packet/filter Hit Ctrl^C to stop recording ^C $ trace-cmd report trace.dat ... <idle>-0 [002] 40398.221702: out_packet: 00070008 9001427a: 058: 31 <idle>-0 [002] 40398.221703: in_packet: 01020010 9001ffff: 002: 15 <idle>-0 [002] 40398.221703: out_packet: 00070010 9001ffff: 002: 32 <idle>-0 [002] 40398.223679: in_packet: 010b0010 900157e4: 090: 16 <idle>-0 [002] 40398.223681: out_packet: 00070010 900157e4: 058: 33 <idle>-0 [002] 40398.223681: in_packet: 010b0018 9001714f: 090: 17 ... One line represent one packet. The legend for the last four fields is: - The first quadlet of CIP header - The second quadlet of CIP header - The number of included quadlets - The index of packet inner buffer maintained by this module Currently, when detecting packet discontinuity, this module stops packet streaming. This is reasonable to packet streaming implementation. However, to identify the quirks, packet streaming should continue to dump enough sequence of packet. This commit adds a condition statement and a marker for this purpose. --- sound/firewire/Makefile | 3 ++ sound/firewire/amdtp-stream-trace.h | 68 +++++++++++++++++++++++++++++++++++++ sound/firewire/amdtp-stream.c | 17 +++++++++- 3 files changed, 87 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/amdtp-stream-trace.h diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 003c090..0ee1fb1 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,3 +1,6 @@ +# To find a header included by define_trace.h. +CFLAGS_amdtp-stream.o := -I$(src) + snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp-stream.o amdtp-am824.o snd-isight-objs := isight.o diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h new file mode 100644 index 0000000..316d495 --- /dev/null +++ b/sound/firewire/amdtp-stream-trace.h @@ -0,0 +1,68 @@ +/* + * amdtp-stream-trace.h - Linux tracing application for ALSA AMDTP engine + * + * Copyright (c) 2016 Takashi Sakamoto + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_firewire_lib + +#if !defined(_AMDTP_STREAM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define _AMDTP_STREAM_TRACE_H + +#include <linux/tracepoint.h> + +TRACE_EVENT(in_packet, + TP_PROTO(u32 cip_header0, u32 cip_header1, unsigned int payload_quadlets, unsigned int index), + TP_ARGS(cip_header0, cip_header1, payload_quadlets, index), + TP_STRUCT__entry( + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->cip_header0 = cip_header0; + __entry->cip_header1 = cip_header1; + __entry->payload_quadlets = payload_quadlets; + __entry->index = index; + ), + TP_printk( + "%08x %08x: %03u: %02u", + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->index) +); + +TRACE_EVENT(out_packet, + TP_PROTO(u32 cip_header0, u32 cip_header1, unsigned int payload_quadlets, unsigned int index), + TP_ARGS(cip_header0, cip_header1, payload_quadlets, index), + TP_STRUCT__entry( + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->cip_header0 = cip_header0; + __entry->cip_header1 = cip_header1; + __entry->payload_quadlets = payload_quadlets; + __entry->index = index; + ), + TP_printk( + "%08x %08x: %03u: %02u", + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->index) +); + +#endif + +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#undef TRACE_INCLUDE_FILE +#define TRACE_INCLUDE_FILE amdtp-stream-trace +#include <trace/define_trace.h> diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ed29026..152375f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -15,6 +15,10 @@ #include <sound/pcm_params.h> #include "amdtp-stream.h" +/* Always support tracepoints enabled by Linux tracing subsystem. */ +#define CREATE_TRACE_POINTS +#include "amdtp-stream-trace.h" + #define TICKS_PER_CYCLE 3072 #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) @@ -426,6 +430,10 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; + + trace_out_packet(be32_to_cpu(buffer[0]), be32_to_cpu(buffer[1]), + payload_length / 4, s->packet_index); + if (queue_out_packet(s, payload_length, false) < 0) return -EIO; @@ -451,6 +459,9 @@ static int handle_in_packet(struct amdtp_stream *s, cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); + trace_in_packet(cip_header[0], cip_header[1], payload_quadlets, + s->packet_index); + /* * This module supports 'Two-quadlet CIP header with SYT field'. * For convenience, also check FMT field is AM824 or not. @@ -523,7 +534,11 @@ static int handle_in_packet(struct amdtp_stream *s, dev_err(&s->unit->device, "Detect discontinuity of CIP: %02X %02X\n", s->data_block_counter, data_block_counter); - return -EIO; + if (!trace_in_packet_enabled()) + return -EIO; + + /* To identifying this situation. */ + trace_in_packet(0xffffffff, 0xffffffff, 999, 99); } pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt); -- 2.7.3

3 5

[alsa-devel] [PATCH] ASoC: rt5677: Avoid duplicate the same test in each switch case
by Axel Lin 30 Mar '16

30 Mar '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 24 +++++------------------- 1 file changed, 5 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 33e290b..b3f1db5 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1241,60 +1241,46 @@ static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source, regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >> RT5677_AD_STO1_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 10: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >> RT5677_AD_STO2_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 9: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >> RT5677_AD_STO3_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 8: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >> RT5677_AD_STO4_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 7: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >> RT5677_AD_MONOL_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 6: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >> RT5677_AD_MONOR_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; default: - break; + return 0; } + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + return 0; } -- 2.1.4

2 1

[alsa-devel] Applied "ASoC: wm8960: Depends on I2C" to the asoc tree
by Mark Brown 30 Mar '16

30 Mar '16

The patch ASoC: wm8960: Depends on I2C 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 3fcdfc9dad07c2a6ea19dfb98a2151cd86d06c6a Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 30 Mar 2016 11:02:49 -0700 Subject: [PATCH] ASoC: wm8960: Depends on I2C Now that this is directly user selectable it needs to care about its dependencies. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 196101b2eab5..3ebf29bc87d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -910,6 +910,7 @@ config SND_SOC_WM8955 config SND_SOC_WM8960 tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate -- 2.8.0.rc3

1 0

[alsa-devel] [PATCH 0/2] ALSA: bebob/fireworks: remove module parameters to reduce maintenance work
by Takashi Sakamoto 30 Mar '16

30 Mar '16

Hi, This patchset removes some module parameters from ALSA bebob and fireworks driver. These two modules have three parameters for users to assign preferable number and strings to a sound card as some modules in ALSA kernel land. However, they're not used so widely, and practically I suspect their advantages. To simplify and unify the shape of modules in ALSA firewire stack, I'd like to remove them. Takashi Sakamoto (2): ALSA: bebob: remove module parameters to reduce maintenance work ALSA: fireworks: remove module parameters to reduce maintenance work sound/firewire/bebob/bebob.c | 49 +++++------------------------------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/fireworks/fireworks.c | 43 +++---------------------------- sound/firewire/fireworks/fireworks.h | 1 - 4 files changed, 10 insertions(+), 84 deletions(-) -- 2.7.3

2 4

[alsa-devel] [PATCH] ASoC: nau8825: reduce standby power consumption
by John Hsu 30 Mar '16

30 Mar '16

Decrease internal clock frequency for power saving when standby. But clock divider needs restore when MCLK as system clock in playback. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..4c8572c 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1116,6 +1116,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + /* MCLK not changed by clock tree */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0); ret = nau8825_mclk_prepare(nau8825, freq); if (ret) return ret; @@ -1126,6 +1129,13 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + /* Decrease the VCO frequency for power saving */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; -- 2.6.4

2 3

[alsa-devel] [PATCH v8 0/6] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 30 Mar '16

30 Mar '16

The only change to v7: Remove the unused variable from tda998x_configure_audio(). Not much have changed since v6[1] patch series. I did everything addressed every comment, mostly by robots :), to previous series and there was one bug fixed in tda998x side. Unless there is any significant reaction to this series, I am going to put it to rest for a while now. I have other things to do. There is currently two other patch series[2][3] that depend on the first two (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 two 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 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 (6): ALSA: pcm: add IEC958 channel status helper for hw_params 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 | 411 +++++++++++++++++++++ 12 files changed, 914 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

5 14

[alsa-devel] [PATCH] ASoC: nau8825: fix interrupt fails and unstable after resume
by John Hsu 30 Mar '16

30 Mar '16

Modify power management function to sync behavior with set bias function. And codec needs to config interrupt setting again to recover interrupt. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 70 ++++++++++++++++++++++++++++------------------ 1 file changed, 43 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 106c391..b8af46b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1204,6 +1204,42 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static void nau8825_configure_interrupt(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interrupts + * It is the same as force_enable_pin("ADC") we do later + */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); +} + +static void nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + nau8825_configure_interrupt(nau8825); + nau8825_restart_jack_detection(regmap); +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1233,6 +1269,8 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + if (nau8825->irq) + nau8825_resume_setup(nau8825); } break; @@ -1346,29 +1384,9 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + nau8825_configure_interrupt(nau8825); ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, @@ -1440,24 +1458,22 @@ static int nau8825_i2c_remove(struct i2c_client *client) #ifdef CONFIG_PM_SLEEP static int nau8825_suspend(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - disable_irq(client->irq); + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); return 0; } static int nau8825_resume(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); + enable_irq(nau8825->irq); return 0; } -- 2.6.4

2 2

[alsa-devel] [PATCH V2 1/1] ASoC: add a core API to share more product information with user space
by han.lu＠intel.com 30 Mar '16

30 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Add a core API to share more machine/board information, for user space apps such as PA and UCM to distinguish various products. Previously on ASoC, the card short name, driver name and long name are all the same as the machine driver name. For example, on T100TA: $ cat /proc/asound/cards 0 [bytcr-rt5640]: bytcr-rt5640 - bytcr-rt5640 bytcr-rt5640 The patch adds more board information: card driver name ---> machine driver name card short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME card long name and card component ---> short name:driver name(:DMI_SYS_VENDOR if available)(:firmware name if available) For example, on T100TA: $ cat /proc/asound/cards 0 [T100TA ]: bytcr-rt5640 - T100TA T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_ sst_0f28.bin $ amixer -c0 info Card hw:0 'T100TA'/'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_ sst_0f28.bin' Mixer name : '' Components : 'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_s st_0f28.bin' Controls : 256 Simple ctrls : 228 Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a21..4e80444 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -486,6 +486,9 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream); int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt); +int snd_soc_set_card_names(struct snd_soc_card *card, const char *board, + const char *vendor, const char *firmware); + /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e7..3162f3e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -152,6 +152,8 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { {} }; +static struct snd_soc_card byt_rt5640_card; + static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; @@ -159,6 +161,19 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_card *card = runtime->card; const struct snd_soc_dapm_route *custom_map; int num_routes; + const char *board, *vendor; + struct sst_acpi_mach *mach = byt_rt5640_card.dev->platform_data; + + /* Add machine/board information for userspace */ + board = dmi_get_system_info(DMI_PRODUCT_NAME); + if (!board) + board = dmi_get_system_info(DMI_BOARD_NAME); + vendor = dmi_get_system_info(DMI_SYS_VENDOR); + ret = snd_soc_set_card_names(card, board, vendor, mach->fw_filename); + if (ret < 0) { + dev_err(card->dev, "unable to set card names\n"); + return ret; + } card->dapm.idle_bias_off = true; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..ab8e7e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1828,6 +1828,58 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); +/** + * snd_soc_set_card_names() - Set the shortname/drivername/longname/component + * of a ASoC card. + * @card: The card to set names + * @board: The board name or product name + * @vendor: The vendor name + * @firmware: The firmware name + * + * This function adds more information for the userspace to distinguish + * different products: + * card driver name ---> machine driver name + * card short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME + * card long name and + * card component ---> short name:driver name(:DMI_SYS_VENDOR) + * (:firmware name) + * + * Returns 0 on success, otherwise a negative error code. + */ +int snd_soc_set_card_names(struct snd_soc_card *card, const char *board, + const char *vendor, const char *firmware) +{ + int ret = 0; + char *name; + + if (!board) { + dev_err(card->dev, "ASoC: the board/product name is empty!\n"); + return -1; + } + /* card driver name */ + card->driver_name = card->name; + /* card short name */ + card->name = board; + /* card long name / card component */ + name = kstrdup(card->name, GFP_KERNEL); + if (!name) + return -ENOMEM; + strcat(name, ":"); + strcat(name, card->driver_name); + strcat(name, ":"); + if (vendor) + strcat(name, vendor); + strcat(name, ":"); + if (firmware) + strcat(name, firmware); + ret = snd_component_add(card->snd_card, name); + card->long_name = card->snd_card->components; + + kfree(name); + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_set_card_names); + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; -- 2.5.0

3 2

[alsa-devel] [PATCH 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 30 Mar '16

30 Mar '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 ++++- include/topology.h | 79 +++++++- src/topology/data.c | 470 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 669 insertions(+), 8 deletions(-) -- 2.5.0

4 14

[alsa-devel] [PATCH] ALSA - hda: hdmi check NULL pointer in hdmi_set_chmap
by libin.yang＠linux.intel.com 30 Mar '16

30 Mar '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

3 2

[alsa-devel] bytcr_rt5640, baytrail: trying to get audio working
by Bernhard Übelacker 30 Mar '16

30 Mar '16

Hello, (This message was sent to alsa-user a week ago, but created no response there. Therefore I hope I am allowed to ask here again.) I am trying to get audio working on a small Intel Z3736F based device. Some more details I have collected in Debian wiki [1]. My tests are done with a current Debian testing. Firmware is from package firmware-intel-sound: sha1sum: 9e7c514b224d674cd3fb1301ce693c42b01f02e0 fw_sst_0f28.bin md5sum: 63765966238afad810b6a568d9b59620 fw_sst_0f28.bin Tests were done on linux v4.5-rc7-244-g03c668a. With a change adjusting the IRQ described in [2]. After a fresh boot I tried the commands below (from [3]). But did not get hearable output, except a "crack" on device open and close. Now I am not sure how to proceed. Am I using the right firmware? Is the amixer sequence ok for this device? Can all the options in alsamixer be tested or is there a risk of damaging the device? Kind regards, Bernhard [1] https://wiki.debian.org/InstallingDebianOn/PIPO/PIPO%20X8 [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094663.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/094080.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094583.html [5] https://sourceforge.net/p/alsa/mailman/alsa-user/thread/56EC0F8A.4010009%40… echo -n 'module snd_soc_sst_mfld_platform +p' > /sys/kernel/debug/dynamic_debug/control #DSP routing amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off #Codec routin, if you have different one please modfiy amixer -c0 sset "DAC MIXL INF1" on amixer -c0 sset "DAC MIXR INF1" on amixer -c0 sset "Stereo DAC MIXL DAC L1" on amixer -c0 sset "Stereo DAC MIXR DAC R1" on amixer -c0 sset "SPK MIXL DAC L1" on amixer -c0 sset "SPK MIXR DAC R1" on amixer -c0 sset "SPOL MIX SPKVOL L" on amixer -c0 sset "SPOR MIX SPKVOL R" on amixer -c0 sset "SPOL MIX DAC L1" on amixer -c0 sset "SPOR MIX DAC R1" on amixer -c0 sset "Speaker L" on amixer -c0 sset "Speaker R" on amixer -c0 sset "Headphone" off #amixer -c0 sset "Ext Spk" on #does not exist amixer -c0 sset "Speaker" on # LANG=C speaker-test --nloops=1 speaker-test 1.1.0 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 16 to 204800 Period size range from 8 to 102400 Using max buffer size 204800 Periods = 4 was set period_size = 51200 was set buffer_size = 204800 0 - Front Left Time per period = 2.109224 # cat /proc/interrupts | grep -E "CPU|sst" CPU0 CPU1 CPU2 CPU3 1: 1 35 0 0 IO-APIC 29-fasteoi intel_sst_driver # LANG=C dmesg -T | grep -i -E "bytcr|snd|rt56|alsa|sst|Baytrail|80860F28:00|SSP|lpe" --color [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: LPE base: 0x90a00000 size:0x200000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: IRAM base: 0x90ac0000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: DRAM base: 0x90b00000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: SHIM base: 0x90b40000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: Mailbox base: 0x90b44000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [Thu Mar 17 11:49:59 2016] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [Thu Mar 17 11:49:59 2016] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [Thu Mar 17 11:49:59 2016] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [Thu Mar 17 11:49:59 2016] bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp2-port mapping ok [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Volume 0, -1440 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Volume 0, 0 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Ramp Delay50 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Volume 0, -1440 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Volume 0, 0 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Ramp Delay50 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Volume 0, -1440 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Mute 1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Volume 0, 0 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Mute 1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Ramp Delay50 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: enable=1, usage=0 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 08 00 ff ff ff ff 55 00 00 00 ............U... [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter:sst_get_ssp_mode, format=4000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter:sst_get_frame_sync_polarity, format=200 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: ssp_id: 3 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 1a 00 ff ff ff ff 75 00 12 00 ............u... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 03 00 03 00 98 00 03 ff 03 ff 03 00 01 00 18 00 ................ [Thu Mar 17 11:51:08 2016] writing to lpe: 00000020: 01 01 .. [Thu Mar 17 11:51:08 2016] intel_sst_acpi 80860F28:00: Alloc for str 1 pipe 0x90 [Thu Mar 17 11:51:08 2016] Baytrail Audio Port: setting buffer ptr param [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=codec_out0 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 01 00 00 1e 00 ff ff ff ff 82 00 16 00 ................ [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 82 00 12 00 03 00 01 02 04 08 10 20 40 80 01 02 ........... @... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000020: 04 08 10 20 40 80 ... @. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=codec_out0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: control name=codec_out0 Gain 0 Volume [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 01 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 00 02 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget = codec_out0 mix 0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: val = 0x2000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: reg: 0x2000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: input id: 0x8dff, nb_inputs: 1 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: ff 02 ff ff 03 00 01 00 ff 8d ff ff ............ [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget = media0_out mix 0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: val = 0x40000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: reg: 0x40000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: input id: 0x90ff, nb_inputs: 1 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 03 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: ff 12 ff ff 03 00 01 00 ff 90 ff ff ............ [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget=media0_out [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: task=3, location=0x12ff [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 03 00 00 0a 00 ff 12 ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 .. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=media0_out [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget=pcm0_in [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: task=1, location=0x8dff [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 0a 00 ff 8d ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 .. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=pcm0_in [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: control name=pcm0_in Gain 0 Volume [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 01 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 00 8d 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: enter, dai-name=media-cpu-dai dir=0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Stream name=Headset Playback [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: send gains for widget=media1_in [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: control name=media1_in Gain 0 Volume [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 03 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 00 90 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:08 2016] Baytrail Audio Port: sst_platform_pcm_trigger called [Thu Mar 17 11:51:08 2016] Baytrail Audio Port: sst: Trigger Start [Thu Mar 17 11:51:08 2016] SSP2-Codec: sst_platform_pcm_trigger called [Thu Mar 17 11:51:10 2016] Baytrail Audio Port: sst_platform_pcm_trigger called [Thu Mar 17 11:51:10 2016] Baytrail Audio Port: sst: in stop [Thu Mar 17 11:51:10 2016] SSP2-Codec: sst_platform_pcm_trigger called [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: enter, dai-name=media-cpu-dai dir=0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Stream name=Headset Playback [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: send gains for widget=media1_in [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: control name=media1_in Gain 0 Volume [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 02 01 03 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 01 00 00 90 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: ssp_id: 3 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 1a 00 ff ff ff ff 75 00 12 00 ............u... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 03 00 00 00 98 00 03 ff 03 ff 03 00 01 00 18 00 ................ [Thu Mar 17 11:51:10 2016] writing to lpe: 00000020: 01 01 .. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: enable=0, usage=1 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 08 00 ff ff ff ff 0e 00 00 00 ................ [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: enter, dai-name=media-cpu-dai dir=0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Stream name=Headset Playback [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: send gains for widget=media1_in [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: control name=media1_in Gain 0 Volume [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 02 01 03 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 01 00 00 90 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:10 2016] intel_sst_acpi 80860F28:00: Free for str 1 pipe 0x90 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget=pcm0_in [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: task=1, location=0x8dff [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 0a 00 ff 8d ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 00 00 .. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget=media0_out [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: task=3, location=0x12ff [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 03 00 00 0a 00 ff 12 ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 00 00 .. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget = codec_out0 mix 0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: val = 0x2000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: reg: 0x2000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: input id: 0x8dff, nb_inputs: 1 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: ff 02 ff ff 00 00 01 00 ff 8d ff ff ............ [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget = media0_out mix 0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: val = 0x40000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: reg: 0x40000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: input id: 0x90ff, nb_inputs: 1 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 03 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: ff 12 ff ff 00 00 01 00 ff 90 ff ff ............ [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=codec_out0

2 3

[alsa-devel] [PATCH 1/5] Sound: SOC: TAS571x: added missing register literals
by Petr Kulhavy 30 Mar '16

30 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to register 0x25 Signed-off-by: Petr Kulhavy <petr(a)barix.com> Tested-by: Petr Kulhavy <petr(a)barix.com> --- sound/soc/codecs/tas571x.c | 4 ++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..346d3da 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: intel: make Kconfig selects depend on ACPI
by Randy Dunlap 29 Mar '16

29 Mar '16

From: Randy Dunlap <rdunlap(a)infradead.org> Prevent SND_SST_IPC_ACPI from being selected when CONFIG_ACPI is not enabled. This fixes the kconfig warning: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/Kconfig | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) --- linux-next-20160308.orig/sound/soc/intel/Kconfig +++ linux-next-20160308/sound/soc/intel/Kconfig @@ -100,7 +100,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH depends on X86 && I2C select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR @@ -113,7 +113,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH depends on X86 && I2C select SND_SOC_RT5651 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR @@ -126,7 +126,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell @@ -139,7 +139,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell @@ -152,7 +152,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI select SND_SOC_MAX98090 select SND_SOC_TS3A227E select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell

2 2

[alsa-devel] [PATCH] ASoC: topology: ABI - Define types for vendor tuples
by mengdong.lin＠linux.intel.com 29 Mar '16

29 Mar '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Tuples, a pair of token and value, can be used to define vendor specific data, for controls and widgets. This can avoid importing binary data blob from other files. Vendor specific tuple arrays will be embeded in the private data buffer of a control or widget object. To be backward compatible, union is used to define the tuple arrays in the existing private data ABI object 'struct snd_soc_tplg_private'. Vendors need to make sure the token values defined by the topology conf file match those defined by their driver. Now supported tuple types are uuid, string, bool, byte, short and word. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index c4cc1e4..db86b447 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -116,6 +116,14 @@ #define SND_SOC_TPLG_STREAM_PLAYBACK 0 #define SND_SOC_TPLG_STREAM_CAPTURE 1 +/* vendor tuple types */ +#define SND_SOC_TPLG_TUPLE_TYPE_UUID 0 +#define SND_SOC_TPLG_TUPLE_TYPE_STRING 1 +#define SND_SOC_TPLG_TUPLE_TYPE_BOOL 2 +#define SND_SOC_TPLG_TUPLE_TYPE_BYTE 3 +#define SND_SOC_TPLG_TUPLE_TYPE_WORD 4 +#define SND_SOC_TPLG_TUPLE_TYPE_SHORT 5 + /* * Block Header. * This header precedes all object and object arrays below. @@ -132,6 +140,35 @@ struct snd_soc_tplg_hdr { __le32 count; /* number of elements in block */ } __attribute__((packed)); +/* vendor tuple for uuid */ +struct snd_soc_tplg_vendor_uuid_elem { + __le32 token; + char uuid[16]; +} __attribute__((packed)); + +/* vendor tuple for a bool/byte/short/word value */ +struct snd_soc_tplg_vendor_value_elem { + __le32 token; + __le32 value; +} __attribute__((packed)); + +/* vendor tuple for string */ +struct snd_soc_tplg_vendor_string_elem { + __le32 token; + char string[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +struct snd_soc_tplg_vendor_array { + __le32 size; /* size in bytes of the array, including all elements */ + __le32 type; /* SND_SOC_TPLG_TUPLE_TYPE_ */ + __le32 num_elems; /* number of elements in array */ + union { + struct snd_soc_tplg_vendor_uuid_elem uuid[0]; + struct snd_soc_tplg_vendor_value_elem value[0]; + struct snd_soc_tplg_vendor_string_elem string[0]; + }; +} __attribute__((packed)); + /* * Private data. * All topology objects may have private data that can be used by the driver or @@ -139,7 +176,10 @@ struct snd_soc_tplg_hdr { */ struct snd_soc_tplg_private { __le32 size; /* in bytes of private data */ - char data[0]; + union { + char data[0]; + struct snd_soc_tplg_vendor_array array[0]; + }; } __attribute__((packed)); /* -- 2.5.0

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm_adsp: Show avail in bytes to match other messages
by Charles Keepax 29 Mar '16

29 Mar '16

All other debug messages talk about data on the compressed stream in bytes except avail which is shown in words. To avoid confusion show avail in bytes as well. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3b1cb1..3c5ef13 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2805,7 +2805,7 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) avail += wm_adsp_buffer_size(buf); adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", - buf->read_index, write_index, avail); + buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); buf->avail = avail; -- 2.1.4

2 8

[alsa-devel] [PATCH 1/2] ASoC: wm8962: Disable clock if wm8962_runtime_resume() fails
by Fabio Estevam 29 Mar '16

29 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> When regulator_bulk_enable() fails inside wm8962_runtime_resume(), we should disable the previously enabled clock. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/wm8962.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8822360..f3f71ba 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3800,7 +3800,7 @@ static int wm8962_runtime_resume(struct device *dev) if (ret != 0) { dev_err(dev, "Failed to enable supplies: %d\n", ret); - return ret; + goto disable_clock; } regcache_cache_only(wm8962->regmap, false); @@ -3838,6 +3838,10 @@ static int wm8962_runtime_resume(struct device *dev) msleep(5); return 0; + +disable_clock: + clk_disable_unprepare(wm8962->pdata.mclk); + return ret; } static int wm8962_runtime_suspend(struct device *dev) -- 1.9.1

3 5

[alsa-devel] [PATCH] ASoC: wm8962: Correct typo, when setting DSPCLK rate
by Charles Keepax 29 Mar '16

29 Mar '16

The variable dspclk holds the rate of the DSPCLK, but the variable sysclk holds an identifier for the clock. Currently if read a non-sensical value from the DSPCLK_DIV register we assign sysclk to dspclk, clearly this was intended to be sysclk_rate. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8822360..720a14e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2471,7 +2471,7 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) break; default: dev_warn(codec->dev, "Unknown DSPCLK divisor read back\n"); - dspclk = wm8962->sysclk; + dspclk = wm8962->sysclk_rate; } dev_dbg(codec->dev, "DSPCLK is %dHz, BCLK %d\n", dspclk, wm8962->bclk); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Adjust clk definitions so that simple card can work
by Fabio Estevam 29 Mar '16

29 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> When trying to use simple card with wm8962 the following probe error happens: wm8962 0-001a: simple-card: set_sysclk error asoc-simple-card sound: ASoC: failed to init 202c000.ssi-wm8962: -22 asoc-simple-card sound: ASoC: failed to instantiate card -22 asoc-simple-card: probe of sound failed with error -22 In simple-card.c the snd_soc_dai_set_sysclk() function is called with clk_id as 0, which is an invalid clock for wm8962. Adjust the clocks source definitions in wm8962.h so that the simple card driver can work successfully. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/wm8962.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index 910aafd..e63a318 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -16,9 +16,9 @@ #include <asm/types.h> #include <sound/soc.h> -#define WM8962_SYSCLK_MCLK 1 -#define WM8962_SYSCLK_FLL 2 -#define WM8962_SYSCLK_PLL3 3 +#define WM8962_SYSCLK_MCLK 0 +#define WM8962_SYSCLK_FLL 1 +#define WM8962_SYSCLK_PLL3 2 #define WM8962_FLL 1 -- 1.9.1

3 2

[alsa-devel] ADAU1761 default register value problem
by Ricard Wanderlof 29 Mar '16

29 Mar '16

I have a problem with the ADAU1761 (although the problem I would think would manifest itself with a veriety of codecs). The register values are cached to minimize the number of writes needed, and there is a set of default values specified. However, this assumes that when the software boots the chip is always in the default state. This is not necessarily the case, because the chip has no external reset line and resets its internal state using an internal power-on-reset circuit, so if the system is rebooted without a power cycle, the regcache will contain the default values of the registers, whereas the actual registers in the chip will retain their previous values. Hence, if attempting to set a value in a register which is in fact identical to the default value, it will not be written, and the value that is currently in the register will remain. This can lead to all sorts of misconfigurations, depending on the previous state of the chip. One way to fix this is simply to avoid setting up a set of default registers (looking at regcache.c, one way seems to be to omit setting the reg_defaults and num_reg_defaults members of the struct regmap_config during initialization, and instead setting num_reg_defaults_raw to the number of registers the chip has, but not setting reg_defaults_raw; I don't know if this is how it is intended to be used however). The regcache framework will then read all the register values from the chip and use them as the new defaults. While this would seem to be technically correct, the question is if this is acceptable? A drawback is that there will be a lot of I2C or SPI accesses just to get the default values. Or should the caching behavior at startup be configured in some way, for instance via devicetree? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH] ALSA: hda - Clear the leftover component assignment at snd_hdac_i915_exit()
by Takashi Iwai 29 Mar '16

29 Mar '16

The commit [d745f5e7b8b2: ALSA: hda - Add the pin / port mapping on Intel ILK and VLV] introduced a WARN_ON() to check the pointer for avoiding the double initializations. But hdac_acomp pointer wasn't cleared at snd_hdac_i915_exit(), thus after reloading the HD-audio driver, it may result in the false positive warning. This patch makes sure to clear the leftover pointer at exit. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=94736 Reported-by: Daniela Doras-prodan <daniela.doras-prodan(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 750a4ea49fa9..ae0f305a7e41 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -372,6 +372,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; return 0; } -- 2.7.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Fix for unloading module only when it is loaded
by jeeja.kp＠intel.com 29 Mar '16

29 Mar '16

From: Jeeja KP <jeeja.kp(a)intel.com> Module needs to be unloaded only when it is loaded successfully. To fix this, first correct the module state sequence and set module state to LOADED if module is loaded successfully. When unloading the module check if module state is not in UNINIT, then unload it. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 5 ++++- sound/soc/intel/skylake/skl-topology.h | 8 ++++---- 2 files changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 545b4e7..3f393cb 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -490,6 +490,8 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) mconfig->id.module_id, mconfig->guid); if (ret < 0) return ret; + + mconfig->m_state = SKL_MODULE_LOADED; } /* update blob if blob is null for be with default value */ @@ -524,7 +526,8 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, list_for_each_entry(w_module, &pipe->w_list, node) { mconfig = w_module->w->priv; - if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod) + if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod && + mconfig->m_state > SKL_MODULE_UNINIT) return ctx->dsp->fw_ops.unload_mod(ctx->dsp, mconfig->id.module_id); } diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index de3c401..d2d9230 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -274,10 +274,10 @@ struct skl_pipe { enum skl_module_state { SKL_MODULE_UNINIT = 0, - SKL_MODULE_INIT_DONE = 1, - SKL_MODULE_LOADED = 2, - SKL_MODULE_UNLOADED = 3, - SKL_MODULE_BIND_DONE = 4 + SKL_MODULE_LOADED = 1, + SKL_MODULE_INIT_DONE = 2, + SKL_MODULE_BIND_DONE = 3, + SKL_MODULE_UNLOADED = 4, }; struct skl_module_cfg { -- 2.5.0

2 3

[alsa-devel] [PATCH] ASoC: nau8825: change output power for interrupt
by John Hsu 29 Mar '16

29 Mar '16

The interrupt clock is gated by x1[10:8], one of them needs to be enabled all the time for interrupts to happen. We change codec to enable ADC because it's helpful to reduce playback pop noise. Don't use force enable pin to enable ADC instead of ADC widget event. That won't interfere DAPM operation and let bias work normally. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 40 ++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 1 + 2 files changed, 29 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 4c8572c..106c391 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -223,6 +223,29 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) } } +static int nau8825_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + break; + case SND_SOC_DAPM_POST_PMD: + if (!nau8825->irq) + regmap_update_bits(nau8825->regmap, + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_ADC, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + static int nau8825_pump_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -338,7 +361,9 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, + nau8825_adc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), @@ -928,13 +953,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* The interrupt clock is gated by x1[10:8], - * one of them needs to be enabled all the time for - * interrupts to happen. - */ - snd_soc_dapm_force_enable_pin(dapm, "DDACR"); - snd_soc_dapm_sync(dapm); - /* Unmask interruptions. Handler uses dapm object so we can enable * interruptions only after dapm is fully initialized. */ @@ -1338,11 +1356,9 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) /* Enable internal VCO needed for interruptions */ nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - /* Enable DDACR needed for interrupts - * It is the same as force_enable_pin("DDACR") we do later - */ + /* Enable ADC needed for interrupts */ regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); /* Chip needs one FSCLK cycle in order to generate interrupts, * as we cannot guarantee one will be provided by the system. Turning diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 5fe009d..4427df9 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -99,6 +99,7 @@ #define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) #define NAU8825_ENABLE_DACL_SFT 9 #define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_ADC (1 << NAU8825_ENABLE_ADC_SFT) #define NAU8825_ENABLE_SAR_SFT 1 /* CLK_DIVIDER (0x3) */ -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: wm8960: Provide a menu selection text
by Fabio Estevam 29 Mar '16

29 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a..196101b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -909,7 +909,7 @@ config SND_SOC_WM8955 tristate config SND_SOC_WM8960 - tristate + tristate "Wolfson Microelectronics WM8960 CODEC" config SND_SOC_WM8961 tristate -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] Asoc: bytcr-rt5640: add more machine/board information for user space
by han.lu＠intel.com 29 Mar '16

29 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Add more machine/board information for PA and UCM. Previously on ASoc, the card short name, driver name and long name are all the same as the machine driver name. On T100TA it looks like: $ cat /proc/asound/cards 0 [bytcr-rt5640]: bytcr-rt5640 - bytcr-rt5640 bytcr-rt5640 The patch made changes below: driver name ---> machine driver name short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME long name ---> short name:driver name(:DMI_SYS_VENDOR if applicable) And add card component: long name:firmware name. On T100TA it looks like: $ cat /proc/asound/cards 0 [T100TA ]: bytcr-rt5640 - T100TA T100TA:bytcr-rt5640:ASUSTek COMPUTER INC. $ amixer -c0 info Card hw:0 'T100TA'/'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.' Mixer name : '' Components : 'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_s st_0f28.bin' Controls : 256 Simple ctrls : 228 Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e7..068cd58 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -152,6 +152,8 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { {} }; +static struct snd_soc_card byt_rt5640_card; + static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; @@ -159,6 +161,34 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_card *card = runtime->card; const struct snd_soc_dapm_route *custom_map; int num_routes; + const char *dmi_name; + char *component; + struct sst_acpi_mach *mach = byt_rt5640_card.dev->platform_data; + + /* Add machine/board information for userspace: + * card driver name ---> machine driver name + * card short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME + * card long name ---> short name:driver name(:DMI_SYS_VENDOR) + */ + card->driver_name = card->name; + dmi_name = dmi_get_system_info(DMI_PRODUCT_NAME); + if (!dmi_name) + dmi_name = dmi_get_system_info(DMI_BOARD_NAME); + if (dmi_name) + card->name = dmi_name; + card->long_name = kstrdup(card->name, GFP_KERNEL); + strcat((char *)card->long_name, ":"); + strcat((char *)card->long_name, card->driver_name); + dmi_name = dmi_get_system_info(DMI_SYS_VENDOR); + if (dmi_name) { + strcat((char *)card->long_name, ":"); + strcat((char *)card->long_name, dmi_name); + } + /* card component ---> long name:firmware name */ + component = kstrdup(card->long_name, GFP_KERNEL); + strcat(component, ":"); + strcat(component, mach->fw_filename); + snd_component_add(card->snd_card, component); card->dapm.idle_bias_off = true; -- 2.5.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by paul.handrigan＠cirrus.com 29 Mar '16

29 Mar '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 54 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1303 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1585 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..d18181c --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,54 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2015 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* GPIO for IRQ */ + int gpio_irq; + + /* irq line */ + int irq; + + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c..44bd911 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -45,6 +45,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -372,6 +373,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712..40f73fa 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -243,6 +244,7 @@ obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..6a29047 --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1303 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 +#define CS35L33_INT_ATTEMPTS 20 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct regmap *cs35l33_get_regmap(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + return cs35l33->regmap; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .get_regmap = cs35l33_get_regmap, + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "release-rate", &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "hd-rm", &val32) >= 0) + hg_config->hd_rm = val32; + if (of_property_read_u32(hg, "ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "ldo-path-disable", &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "ldo-entry-delay", &val32) >= 0) + hg_config->ldo_entry_delay = val32; + hg_config->vp_hg_auto = of_property_read_bool(hg, "vp-hg-auto"); + if (of_property_read_u32(hg, "vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + int i; + + for (i = 0; i < CS35L33_INT_ATTEMPTS; i++) { + /* ack the irq by reading both status registers */ + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit at the exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + break; + + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_err(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_err(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + if (CS35L33_VBSTMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + } + + if (i == CS35L33_INT_ATTEMPTS) { + dev_err(codec->dev, + "ERROR: Interrupt handler exceeded max attpmpts\n"); + return -EINVAL; + } + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = request_threaded_irq(i2c_client->irq, NULL, cs35l33_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs35l33", cs35l33); + if (ret != 0) + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .owner = THIS_MODULE, + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..2f31952 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

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Re: [alsa-devel] writing driver for MOTU PCI 324, 2408 MK-II
by Jef 28 Mar '16

28 Mar '16

Forest Bond <f_a_b <at> softhome.net> writes: > > Hello all, > > I'm new to the list. Joined up because I've decided I want to write a > driver for the Mark of the Unicorn PCI-324 PCI card. This card allows a > variety of MOTU external interfaces to be used. I own a MOTU 2408 MK > II, an 8 channel interface. This (and similar interfaces from MOTU) are > common in the pro audio world. > > I have emailed MOTU tech support, but have not gotten a response. I > have a feeling this will be the case in the future, as well. MOTU has > strong ties with Apple. > > I've searched the list archive, and, from what I can see, noone has > begun work on a driver for these interfaces. It may not be feasible, > given the lack of technical information available (yeah, yeah, I know, > welcome to the wonderful world of linux driver development). If, > however, anyone is aware of an attempt to write a driver, please let me > know. Also, if anyone is interested in pitching in, send me an email. > > I have limited experience writing drivers (well, next to none, > actually), but I'm gonna give it a shot anyway. > > Couple of questions: > > Can released ASIO drivers be used to gather technical information on the > hardware? > > Are there any good techniques for probing the hardware and discovering > necessary technical info? > > Any help would be appreciated. > > Thanks! > Forest > Hello! I know this is an old post, but I was hoping that the sentiment was still alive and strong. I have recently made the transition to Linux as my primary OS, and I, too, am looking to move my audio gear along as well. Has there been any progress on the PCI-324 linux driver development front? Is there an active project in the works, or has this idea been abandoned? Cheers! And thank you. Jef

1 1

[alsa-devel] [PATCH 0/4] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 28 Mar '16

28 Mar '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Jose Abreu (4): drm/i2c/adv7511: Add audio support ARC: axs10x: Update defconfigs so that audio is enabled ASoC: dwc: Add I2S HDMI audio support using custom platform driver ARC: axs10x: Update defconfigs so that I2S is enabled arch/arc/boot/dts/axs10x_mb.dtsi | 20 +- arch/arc/configs/axs101_defconfig | 4 + arch/arc/configs/axs103_defconfig | 6 + arch/arc/configs/axs103_smp_defconfig | 6 + drivers/gpu/drm/i2c/Kconfig | 8 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 --------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 ++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 384 ++++++++++++- 13 files changed, 1771 insertions(+), 1041 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c -- 1.9.1

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[alsa-devel] [PATCH 0/2] ALSA: dice: fix a fatal bug when no sound card is registered
by Takashi Sakamoto 28 Mar '16

28 Mar '16

Hi, This patchset is for Linux 4.6-rc2. ALSA dice driver causes kernel NULL pointer dereference in a case that no sound card instance is registered and unit is going to be removed. This is observed when dice-based units are connected to Linux system and disconnected within 2 seconds, thus it's rare. Additionally, even if the bug is fixed, the driver dumps warnings for most dice-based models which supports a pair of isochronous streams. This can occur regardless of the former bug. Users can see this for unsupported dice-based models. This patchset fixes these bugs. Takashi Sakamoto (2): ALSA: dice: fix NULL pointer dereference at remove units when sound card is not registered ALSA: firewire-lib: skip releasing stream data when it's not initialized sound/firewire/amdtp-stream.c | 4 ++++ sound/firewire/dice/dice-stream.c | 14 ++++---------- 2 files changed, 8 insertions(+), 10 deletions(-) -- 2.7.3

2 5

[alsa-devel] [PATCH 0/4] ALSA: firewire: some trivial fixes
by Takashi Sakamoto 28 Mar '16

28 Mar '16

Hi, This patchset includes some trivial fixes for ALSA firewire stack. I don't mind to postpone them to 4.7 instead of 4.6-rc2. Regards Takashi Sakamoto (4): ALSA: bebob: remove needless argument from local function ALSA: oxfw: remove needless member from private structure ALSA: fireworks: move model quirk detection code to information parser ALSA: firewire-tascam: add Kconfig entry for TASCAM FW-1804 sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 4 ++-- sound/firewire/fireworks/fireworks.c | 21 +++++++++++---------- sound/firewire/oxfw/oxfw.h | 1 - 4 files changed, 14 insertions(+), 13 deletions(-) -- 2.7.3

2 5

[alsa-devel] [PATCH v2] ALSA: dice: fix memory leak when unplugging
by Takashi Sakamoto 28 Mar '16

28 Mar '16

When sound card is going to be released, dice private data is also released. Then all of data should be released. However, stream data is not released. This causes memory leak when unplugging dice unit. This commit fixes the bug. Fixes: 4bdc495c87b3('ALSA: dice: handle several PCM substreams when any isochronous streams are available') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 845d5e5..ec4db3a 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -446,18 +446,12 @@ end: void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { - struct reg_params tx_params, rx_params; - - snd_dice_transaction_clear_enable(dice); + unsigned int i; - if (get_register_params(dice, &tx_params, &rx_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, &tx_params); - stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); + for (i = 0; i < MAX_STREAMS; i++) { + destroy_stream(dice, AMDTP_IN_STREAM, i); + destroy_stream(dice, AMDTP_OUT_STREAM, i); } - - release_resources(dice); - - dice->substreams_counter = 0; } void snd_dice_stream_update_duplex(struct snd_dice *dice) -- 2.7.3

2 1

[alsa-devel] [PATCH] ALSA: dice: fix memory leak when unplugging
by Takashi Sakamoto 28 Mar '16

28 Mar '16

When sound card is going to be released, dice private data is also released. Then all of data should be released. However, stream data is not released. This causes memory leak when unplugging dice unit. This commit fixes the bug. Fixes: 4bdc495c87b3('ALSA: dice: handle several PCM substreams when any isochronous streams are available') --- sound/firewire/dice/dice-stream.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 845d5e5..ec4db3a 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -446,18 +446,12 @@ end: void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { - struct reg_params tx_params, rx_params; - - snd_dice_transaction_clear_enable(dice); + unsigned int i; - if (get_register_params(dice, &tx_params, &rx_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, &tx_params); - stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); + for (i = 0; i < MAX_STREAMS; i++) { + destroy_stream(dice, AMDTP_IN_STREAM, i); + destroy_stream(dice, AMDTP_OUT_STREAM, i); } - - release_resources(dice); - - dice->substreams_counter = 0; } void snd_dice_stream_update_duplex(struct snd_dice *dice) -- 2.7.3

1 1

Re: [alsa-devel] ASoC: qcom: add mic support
by Dan Carpenter 25 Mar '16

25 Mar '16

Hello Srinivas Kandagatla, The patch fb5d11524eda: "ASoC: qcom: add mic support" from Feb 11, 2016, leads to the following static checker warning: sound/soc/qcom/lpass-platform.c:555 lpass_platform_pcm_new() error: uninitialized symbol 'ret'. sound/soc/qcom/lpass-platform.c 515 csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 516 if (csubstream) { 517 if (v->alloc_dma_channel) 518 data->wrdma_ch = v->alloc_dma_channel(drvdata, 519 SNDRV_PCM_STREAM_CAPTURE); 520 521 if (IS_ERR_VALUE(data->wrdma_ch)) 522 goto capture_alloc_err; ->wrdma_ch is an int so this should just be: if (data->wrdma_ch < 0) { ret = data->wrdma_ch; goto capture_alloc_err; } The other ->alloc_dma_channel uses the same IS_ERR_VALUE() check as well so we should probably clean that up as well. The IS_ERR_VALUE() function is for places which return unsigned long basically. No need to be fancy for normal "negative returns are error" code. 523 524 drvdata->substream[data->wrdma_ch] = csubstream; 525 526 ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, 527 soc_runtime->platform->dev, 528 size, &csubstream->dma_buffer); 529 if (ret) 530 goto capture_alloc_err; 531 532 ret = regmap_write(drvdata->lpaif_map, 533 LPAIF_WRDMACTL_REG(v, data->wrdma_ch), 0); 534 if (ret) { 535 dev_err(soc_runtime->dev, 536 "%s() error writing to wrdmactl reg: %d\n", 537 __func__, ret); 538 goto capture_reg_err; 539 } 540 } 541 542 return 0; 543 544 capture_reg_err: 545 if (csubstream) 546 snd_dma_free_pages(&csubstream->dma_buffer); 547 548 capture_alloc_err: 549 if (psubstream) 550 snd_dma_free_pages(&psubstream->dma_buffer); 551 552 playback_alloc_err: 553 dev_err(soc_runtime->dev, "Cannot allocate buffer(s)\n"); 554 555 return ret; 556 } regards, dan carpenter

1 0

[alsa-devel] [RFC PATCH v2 0/3] start to support MST
by libin.yang＠linux.intel.com 25 Mar '16

25 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> These patches are draft patches to support MST. The patches doesn't work because of lacking support in gfx driver. Libin Yang (3): ALSA: hda - add DP mst verb support ALSA - hda: add devid support in acom ALSA - hda: add DP MST support drivers/gpu/drm/i915/intel_audio.c | 2 +- include/drm/i915_component.h | 2 +- include/sound/hda_i915.h | 6 +- sound/hda/hdac_i915.c | 7 +- sound/pci/hda/hda_codec.c | 86 ++++++++++++++++- sound/pci/hda/hda_codec.h | 3 + sound/pci/hda/patch_hdmi.c | 184 +++++++++++++++++++++++++++---------- 7 files changed, 227 insertions(+), 63 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: adau1701: Reset codec based on sample rate changes
by pascal.huerst＠gmail.com 24 Mar '16

24 Mar '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Instead of checking if mclk/lrclk ratio has changed, check if sample rate has changed. In certain cases, the mclk might be changed in the machine driver, which can lead to the same mclk/lrclk ration, eventhow the sample rate has changed. Since the codec has to be programmed differently for every sample rate, its better to check for samplerate changes instead of mclk/lrclk ration changes. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/adau1701.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 202dea1..1dbbcda 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -111,6 +111,7 @@ struct adau1701 { unsigned int dai_fmt; unsigned int pll_clkdiv; unsigned int sysclk; + unsigned int current_rate; struct regmap *regmap; struct i2c_client *client; u8 pin_config[12]; @@ -436,22 +437,24 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); - unsigned int clkdiv = adau1701->sysclk / params_rate(params); + unsigned int rate = params_rate(params); + unsigned int clkdiv = adau1701->sysclk / rate; unsigned int val; int ret; /* - * If the mclk/lrclk ratio changes, the chip needs updated PLL + * If the sample rate changes, the chip needs updated PLL * mode GPIO settings, and a full reset cycle, including a new * firmware upload. */ - if (clkdiv != adau1701->pll_clkdiv) { - ret = adau1701_reset(codec, clkdiv, params_rate(params)); + if (adau1701->current_rate != rate) { + adau1701->current_rate = rate; + ret = adau1701_reset(codec, clkdiv, rate); if (ret < 0) return ret; } - switch (params_rate(params)) { + switch (rate) { case 192000: val = ADAU1701_DSPCTRL_SR_192; break; -- 2.5.0

4 5

[alsa-devel] [PATCH - tinycompress - brace issue 1/1] tinycompress: fix compilation error: missing brace issue
by francois.gaffie＠intel.com 24 Mar '16

24 Mar '16

From: François Gaffie <francois.gaffie(a)intel.com> Signed-off-by: François Gaffie <francois.gaffie(a)intel.com> diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 68626a4..d35801b 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -273,4 +273,7 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} +#endif #endif diff --git a/include/tinycompress/tinymp3.h b/include/tinycompress/tinymp3.h index a709c39..7015320 100644 --- a/include/tinycompress/tinymp3.h +++ b/include/tinycompress/tinymp3.h @@ -102,5 +102,7 @@ enum mp3_stereo_mode { DUAL = 0x02, MONO = 0x03 }; - +#if defined(__cplusplus) +} +#endif #endif -- 2.5.0 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 0

[alsa-devel] [PATCH 4/4] [meida] media-device: dynamically allocate struct media_devnode
by Mauro Carvalho Chehab 24 Mar '16

24 Mar '16

struct media_devnode is currently embedded at struct media_device. While this works fine during normal usage, it leads to a race condition during devnode unregister. the problem is that drivers assume that, after calling media_device_unregister(), the struct that contains media_device can be freed. This is not true, as it can't be freed until userspace closes all opened /dev/media devnodes. In other words, if the media devnode is still open, and media_device gets freed, any call to an ioctl will make the core to try to access struct media_device, with will cause an use-after-free and even GPF. Fix this by dynamically allocating the struct media_devnode and only freeing it when it is safe. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- drivers/media/media-device.c | 38 ++++++++++++++++++++++------------ drivers/media/media-devnode.c | 7 ++++++- drivers/media/usb/au0828/au0828-core.c | 4 ++-- drivers/media/usb/uvc/uvc_driver.c | 2 +- include/media/media-device.h | 5 +---- include/media/media-devnode.h | 15 ++++++++++++-- sound/usb/media.c | 8 +++---- 7 files changed, 52 insertions(+), 27 deletions(-) diff --git a/drivers/media/media-device.c b/drivers/media/media-device.c index 10cc4766de10..d10dc615e7a8 100644 --- a/drivers/media/media-device.c +++ b/drivers/media/media-device.c @@ -428,7 +428,7 @@ static long media_device_ioctl(struct file *filp, unsigned int cmd, unsigned long arg) { struct media_devnode *devnode = media_devnode_data(filp); - struct media_device *dev = to_media_device(devnode); + struct media_device *dev = devnode->media_dev; long ret; switch (cmd) { @@ -504,7 +504,7 @@ static long media_device_compat_ioctl(struct file *filp, unsigned int cmd, unsigned long arg) { struct media_devnode *devnode = media_devnode_data(filp); - struct media_device *dev = to_media_device(devnode); + struct media_device *dev = devnode->media_dev; long ret; switch (cmd) { @@ -540,7 +540,8 @@ static const struct media_file_operations media_device_fops = { static ssize_t show_model(struct device *cd, struct device_attribute *attr, char *buf) { - struct media_device *mdev = to_media_device(to_media_devnode(cd)); + struct media_devnode *devnode = to_media_devnode(cd); + struct media_device *mdev = devnode->media_dev; return sprintf(buf, "%.*s\n", (int)sizeof(mdev->model), mdev->model); } @@ -718,25 +719,36 @@ EXPORT_SYMBOL_GPL(media_device_cleanup); int __must_check __media_device_register(struct media_device *mdev, struct module *owner) { + struct media_devnode *devnode; int ret; mutex_lock(&mdev->graph_mutex); + devnode = kzalloc(sizeof(*devnode), GFP_KERNEL); + if (!devnode) + return -ENOMEM; + /* Register the device node. */ - mdev->devnode.fops = &media_device_fops; - mdev->devnode.parent = mdev->dev; - mdev->devnode.release = media_device_release; + mdev->devnode = devnode; + devnode->fops = &media_device_fops; + devnode->parent = mdev->dev; + devnode->release = media_device_release; /* Set version 0 to indicate user-space that the graph is static */ mdev->topology_version = 0; - ret = media_devnode_register(&mdev->devnode, owner); - if (ret < 0) + ret = media_devnode_register(mdev, devnode, owner); + if (ret < 0) { + mdev->devnode = NULL; + kfree(devnode); goto err; + } - ret = device_create_file(&mdev->devnode.dev, &dev_attr_model); + ret = device_create_file(&devnode->dev, &dev_attr_model); if (ret < 0) { - media_devnode_unregister(&mdev->devnode); + mdev->devnode = NULL; + media_devnode_unregister(devnode); + kfree(devnode); goto err; } @@ -800,9 +812,9 @@ static void __media_device_unregister(struct media_device *mdev) } /* Check if mdev devnode was registered */ - if (media_devnode_is_registered(&mdev->devnode)) { - device_remove_file(&mdev->devnode.dev, &dev_attr_model); - media_devnode_unregister(&mdev->devnode); + if (media_devnode_is_registered(mdev->devnode)) { + device_remove_file(&mdev->devnode->dev, &dev_attr_model); + media_devnode_unregister(mdev->devnode); } dev_dbg(mdev->dev, "Media device unregistered\n"); diff --git a/drivers/media/media-devnode.c b/drivers/media/media-devnode.c index ae2bc0b7a368..db47063d8801 100644 --- a/drivers/media/media-devnode.c +++ b/drivers/media/media-devnode.c @@ -44,6 +44,7 @@ #include <linux/uaccess.h> #include <media/media-devnode.h> +#include <media/media-device.h> #define MEDIA_NUM_DEVICES 256 #define MEDIA_NAME "media" @@ -74,6 +75,8 @@ static void media_devnode_release(struct device *cd) /* Release media_devnode and perform other cleanups as needed. */ if (devnode->release) devnode->release(devnode); + + kfree(devnode); } static struct bus_type media_bus_type = { @@ -218,7 +221,8 @@ static const struct file_operations media_devnode_fops = { .llseek = no_llseek, }; -int __must_check media_devnode_register(struct media_devnode *devnode, +int __must_check media_devnode_register(struct media_device *mdev, + struct media_devnode *devnode, struct module *owner) { int minor; @@ -237,6 +241,7 @@ int __must_check media_devnode_register(struct media_devnode *devnode, mutex_unlock(&media_devnode_lock); devnode->minor = minor; + devnode->media_dev = mdev; /* Part 2: Initialize and register the character device */ cdev_init(&devnode->cdev, &media_devnode_fops); diff --git a/drivers/media/usb/au0828/au0828-core.c b/drivers/media/usb/au0828/au0828-core.c index 85c13ca5178f..598a85388d77 100644 --- a/drivers/media/usb/au0828/au0828-core.c +++ b/drivers/media/usb/au0828/au0828-core.c @@ -142,7 +142,7 @@ static void au0828_unregister_media_device(struct au0828_dev *dev) struct media_device *mdev = dev->media_dev; struct media_entity_notify *notify, *nextp; - if (!mdev || !media_devnode_is_registered(&mdev->devnode)) + if (!mdev || !media_devnode_is_registered(mdev->devnode)) return; /* Remove au0828 entity_notify callbacks */ @@ -480,7 +480,7 @@ static int au0828_media_device_register(struct au0828_dev *dev, if (!dev->media_dev) return 0; - if (!media_devnode_is_registered(&dev->media_dev->devnode)) { + if (!media_devnode_is_registered(dev->media_dev->devnode)) { /* register media device */ ret = media_device_register(dev->media_dev); diff --git a/drivers/media/usb/uvc/uvc_driver.c b/drivers/media/usb/uvc/uvc_driver.c index 451e84e962e2..302e284a95eb 100644 --- a/drivers/media/usb/uvc/uvc_driver.c +++ b/drivers/media/usb/uvc/uvc_driver.c @@ -1674,7 +1674,7 @@ static void uvc_delete(struct uvc_device *dev) if (dev->vdev.dev) v4l2_device_unregister(&dev->vdev); #ifdef CONFIG_MEDIA_CONTROLLER - if (media_devnode_is_registered(&dev->mdev.devnode)) + if (media_devnode_is_registered(dev->mdev.devnode)) media_device_unregister(&dev->mdev); media_device_cleanup(&dev->mdev); #endif diff --git a/include/media/media-device.h b/include/media/media-device.h index e59772ed8494..b04cfa907350 100644 --- a/include/media/media-device.h +++ b/include/media/media-device.h @@ -347,7 +347,7 @@ struct media_entity_notify { struct media_device { /* dev->driver_data points to this struct. */ struct device *dev; - struct media_devnode devnode; + struct media_devnode *devnode; char model[32]; char driver_name[32]; @@ -403,9 +403,6 @@ struct usb_device; #define MEDIA_DEV_NOTIFY_PRE_LINK_CH 0 #define MEDIA_DEV_NOTIFY_POST_LINK_CH 1 -/* media_devnode to media_device */ -#define to_media_device(node) container_of(node, struct media_device, devnode) - /** * media_entity_enum_init - Initialise an entity enumeration * diff --git a/include/media/media-devnode.h b/include/media/media-devnode.h index e1d5af077adb..cc2b3155593c 100644 --- a/include/media/media-devnode.h +++ b/include/media/media-devnode.h @@ -33,6 +33,8 @@ #include <linux/device.h> #include <linux/cdev.h> +struct media_device; + /* * Flag to mark the media_devnode struct as registered. Drivers must not touch * this flag directly, it will be set and cleared by media_devnode_register and @@ -81,6 +83,8 @@ struct media_file_operations { * before registering the node. */ struct media_devnode { + struct media_device *media_dev; + /* device ops */ const struct media_file_operations *fops; @@ -103,7 +107,8 @@ struct media_devnode { /** * media_devnode_register - register a media device node * - * @devnode: media device node structure we want to register + * @media_dev: struct media_device we want to register a device node + * @devnode: the device node to unregister * @owner: should be filled with %THIS_MODULE * * The registration code assigns minor numbers and registers the new device node @@ -116,7 +121,8 @@ struct media_devnode { * the media_devnode structure is *not* called, so the caller is responsible for * freeing any data. */ -int __must_check media_devnode_register(struct media_devnode *devnode, +int __must_check media_devnode_register(struct media_device *mdev, + struct media_devnode *devnode, struct module *owner); /** @@ -146,9 +152,14 @@ static inline struct media_devnode *media_devnode_data(struct file *filp) * false otherwise. * * @devnode: pointer to struct &media_devnode. + * + * Note: If mdev is NULL, it also returns false. */ static inline int media_devnode_is_registered(struct media_devnode *devnode) { + if (!devnode) + return false; + return test_bit(MEDIA_FLAG_REGISTERED, &devnode->flags); } diff --git a/sound/usb/media.c b/sound/usb/media.c index 6db4878045e5..8fed0adec9e1 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -136,7 +136,7 @@ void media_snd_stream_delete(struct snd_usb_substream *subs) struct media_device *mdev; mdev = mctl->media_dev; - if (mdev && media_devnode_is_registered(&mdev->devnode)) { + if (mdev && media_devnode_is_registered(mdev->devnode)) { media_devnode_remove(mctl->intf_devnode); media_device_unregister_entity(&mctl->media_entity); media_entity_cleanup(&mctl->media_entity); @@ -241,14 +241,14 @@ static void media_snd_mixer_delete(struct snd_usb_audio *chip) if (!mixer->media_mixer_ctl) continue; - if (media_devnode_is_registered(&mdev->devnode)) { + if (media_devnode_is_registered(mdev->devnode)) { media_device_unregister_entity(&mctl->media_entity); media_entity_cleanup(&mctl->media_entity); } kfree(mctl); mixer->media_mixer_ctl = NULL; } - if (media_devnode_is_registered(&mdev->devnode)) + if (media_devnode_is_registered(mdev->devnode)) media_devnode_remove(chip->ctl_intf_media_devnode); chip->ctl_intf_media_devnode = NULL; } @@ -268,7 +268,7 @@ int media_snd_device_create(struct snd_usb_audio *chip, if (!mdev->dev) media_device_usb_init(mdev, usbdev, NULL); - if (!media_devnode_is_registered(&mdev->devnode)) { + if (!media_devnode_is_registered(mdev->devnode)) { ret = media_device_register(mdev); if (ret) { dev_err(&usbdev->dev, -- 2.5.5

2 2

[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 24 Mar '16

24 Mar '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/max98371.c | 428 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 63 +++ 3 files changed, 508 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..1aba140 --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,428 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const voltage_level_text[] = { + "9.5dB", "10.5dB", "11.5dB", "11.5dB", "12.5dB", + "13.5dB", "14.5dB", "15.5dB", "16.5dB", "17.5dB", + "18.5dB", "19.5dB", "20.5dB" +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_voltage, MAX98371_GAIN, 0, + voltage_level_text); + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const unsigned int max98371_gain_tlv[] = { + 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), + 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), + 11, 15, TLV_DB_SCALE_ITEM(0, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Gain", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Voltage Level", max98371_voltage), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + pr_info("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + pr_info("%s: ratio unsupported %d", __func__, blr_clk_ratio); + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + pr_info("%s: rate unsupported %d", __func__, rate); + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", + "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Enable", + "HiFi Playback", MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", + MAX98371_GLOBAL_ENABLE, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC Enable", NULL, "DAI_OUT"}, + {"SPK_OUT", NULL, "DAC Enable"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static int max98371_probe(struct snd_soc_codec *codec) +{ + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + + max98371->codec = codec; + return 0; +} + +static struct snd_soc_codec_driver max98371_codec = { + .probe = max98371_probe, + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..644316a --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,63 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

4 11

[alsa-devel] ASoC: TLV320AIC3x how to reduce alsa playback/capture latency?
by Timur Karaldin 24 Mar '16

24 Mar '16

Hello, I have an issue to reduce playback/capture audio latency for AIC3106. Now I'm using following strategy for initial of alsa backend driver in user level application: snd_pcm_hw_params_set_period_size(...,frame_size,...); snd_pcm_hw_params_set_periods_min(...,&periodsp,...); if(periodsp< user_nperiods) periodsp = user_nperiods snd_pcm_hw_params_set_periods_near(....,&periodsp,...); snd_pcm_hw_set_buffer_size(...,frame_size*periodsp,....); [...] snd_pcm_sw_params_set_start_threshold(..., 0U); snd_pcm_sw_params_set_stop_threshold(..., frame_size*periodsp); if(playback) { availmin=frame_size*(periodsp-user_nperiods+1); }else { availmin=frame_size; } snd_pcm_sw_params_set_avail_min(...,availmin) [...] On far end I hear an echo with delay about 1 second, I definitely know that it is a playback/capture (acoustic echo), and delay is too much for any AEC because of (as I think) low latency in alsa driver or misunderstanding how it should be initialized. I already exclude other delays (jitter delay, network delay) My initial parameters: frame_size = 640, rate = 16000, user_nperiods = 2, but after calling snd_pcm_hw_params_set_periods_min I got minimum periodsp = 16. For these parameter buffer (16 buffers 40ms each) total length of buffer is 0.64 sec. My question is how could I reduce this latency? May be some improvements which allow me to set up less then 16 buffers, or I did wrong initialization or is there any another internal buffer which make such long delay for echo? Cheers, Tim Karaldin

1 0

[alsa-devel] question about cross compilation with alsa
by Alvarado, Claudia 23 Mar '16

23 Mar '16

Hello! I have a general question. We are cross compiling on a Linux machine for an ARM processor on an embedded device. When compiling, we specify paths for the include files and our cross compiled Alsa library. We then try to port this to our embedded device. When trying to run the executable on the embedded device we get two problems: 1. it doesn’t recognize the PCM device called ‘default’— I think we just need to call this out with hw:0,0 but we need to play with this. 2. We get this error: “ALSA lib conf.c(snd_config_update_r)Cannot access file /path/to/alsa.conf" * The path that it lists is the path to our cross compiled alsa lib on our linux box… so it seems that in the compilation process, this path is preserved and transferred to the executable somehow. So my question is, is there a way to specify two different library paths for the two devices (linux and embedded) while compiling so that when we port to the embedded device, the executable knows where to look for the alsa library? Normally, this wouldn’t be an issue because the default on a linux machine is usr/include, which I believe gets searched by default. However, on our embedded device, the path is different. Thanks in advance!

1 0

[alsa-devel] [PATCH v2] [media] media-device: use kref for media_device instance
by Mauro Carvalho Chehab 23 Mar '16

23 Mar '16

Now that the media_device can be used by multiple drivers, via devres, we need to be sure that it will be dropped only when all drivers stop using it. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- v2: The kref is now used only when media_device is allocated via the media_device*_devress. This warrants that other drivers won't be affected, and that we can keep media_device_cleanup() balanced with media_device_init(). drivers/media/media-device.c | 117 +++++++++++++++++++++++++-------- drivers/media/usb/au0828/au0828-core.c | 3 +- include/media/media-device.h | 28 ++++++++ sound/usb/media.c | 3 +- 4 files changed, 118 insertions(+), 33 deletions(-) diff --git a/drivers/media/media-device.c b/drivers/media/media-device.c index c32fa15cc76e..4a97d92a7e7d 100644 --- a/drivers/media/media-device.c +++ b/drivers/media/media-device.c @@ -707,11 +707,16 @@ void media_device_init(struct media_device *mdev) } EXPORT_SYMBOL_GPL(media_device_init); -void media_device_cleanup(struct media_device *mdev) +static void __media_device_cleanup(struct media_device *mdev) { ida_destroy(&mdev->entity_internal_idx); mdev->entity_internal_idx_max = 0; media_entity_graph_walk_cleanup(&mdev->pm_count_walk); +} + +void media_device_cleanup(struct media_device *mdev) +{ + __media_device_cleanup(mdev); mutex_destroy(&mdev->graph_mutex); } EXPORT_SYMBOL_GPL(media_device_cleanup); @@ -721,6 +726,9 @@ int __must_check __media_device_register(struct media_device *mdev, { int ret; + /* Check if mdev was ever registered at all */ + mutex_lock(&mdev->graph_mutex); + /* Register the device node. */ mdev->devnode.fops = &media_device_fops; mdev->devnode.parent = mdev->dev; @@ -731,17 +739,19 @@ int __must_check __media_device_register(struct media_device *mdev, ret = media_devnode_register(&mdev->devnode, owner); if (ret < 0) - return ret; + goto err; ret = device_create_file(&mdev->devnode.dev, &dev_attr_model); if (ret < 0) { media_devnode_unregister(&mdev->devnode); - return ret; + goto err; } dev_dbg(mdev->dev, "Media device registered\n"); - return 0; +err: + mutex_unlock(&mdev->graph_mutex); + return ret; } EXPORT_SYMBOL_GPL(__media_device_register); @@ -773,24 +783,13 @@ void media_device_unregister_entity_notify(struct media_device *mdev, } EXPORT_SYMBOL_GPL(media_device_unregister_entity_notify); -void media_device_unregister(struct media_device *mdev) +static void __media_device_unregister(struct media_device *mdev) { struct media_entity *entity; struct media_entity *next; struct media_interface *intf, *tmp_intf; struct media_entity_notify *notify, *nextp; - if (mdev == NULL) - return; - - mutex_lock(&mdev->graph_mutex); - - /* Check if mdev was ever registered at all */ - if (!media_devnode_is_registered(&mdev->devnode)) { - mutex_unlock(&mdev->graph_mutex); - return; - } - /* Remove all entities from the media device */ list_for_each_entry_safe(entity, next, &mdev->entities, graph_obj.list) __media_device_unregister_entity(entity); @@ -807,38 +806,98 @@ void media_device_unregister(struct media_device *mdev) kfree(intf); } - mutex_unlock(&mdev->graph_mutex); - - device_remove_file(&mdev->devnode.dev, &dev_attr_model); - media_devnode_unregister(&mdev->devnode); + /* Check if mdev devnode was registered */ + if (media_devnode_is_registered(&mdev->devnode)) { + device_remove_file(&mdev->devnode.dev, &dev_attr_model); + media_devnode_unregister(&mdev->devnode); + } dev_dbg(mdev->dev, "Media device unregistered\n"); } + +void media_device_unregister(struct media_device *mdev) +{ + if (mdev == NULL) + return; + + mutex_lock(&mdev->graph_mutex); + __media_device_unregister(mdev); + mutex_unlock(&mdev->graph_mutex); +} EXPORT_SYMBOL_GPL(media_device_unregister); static void media_device_release_devres(struct device *dev, void *res) { } -struct media_device *media_device_get_devres(struct device *dev) +static void do_media_device_unregister_devres(struct kref *kref) { + struct media_device_devres *mdev_devres; struct media_device *mdev; + int ret; - mdev = devres_find(dev, media_device_release_devres, NULL, NULL); - if (mdev) - return mdev; + mdev_devres = container_of(kref, struct media_device_devres, kref); - mdev = devres_alloc(media_device_release_devres, - sizeof(struct media_device), GFP_KERNEL); - if (!mdev) + if (!mdev_devres) + return; + + mdev = &mdev_devres->mdev; + + mutex_lock(&mdev->graph_mutex); + __media_device_unregister(mdev); + __media_device_cleanup(mdev); + mutex_unlock(&mdev->graph_mutex); + mutex_destroy(&mdev->graph_mutex); + + ret = devres_destroy(mdev->dev, media_device_release_devres, + NULL, NULL); + pr_debug("%s: devres_destroy() returned %d\n", __func__, ret); +} + +void media_device_unregister_devres(struct media_device *mdev) +{ + struct media_device_devres *mdev_devres; + + mdev_devres = container_of(mdev, struct media_device_devres, mdev); + kref_put(&mdev_devres->kref, do_media_device_unregister_devres); +} +EXPORT_SYMBOL_GPL(media_device_unregister_devres); + +struct media_device *media_device_get_devres(struct device *dev) +{ + struct media_device_devres *mdev_devres, *ptr; + + mdev_devres = devres_find(dev, media_device_release_devres, NULL, NULL); + if (mdev_devres) { + kref_get(&mdev_devres->kref); + return &mdev_devres->mdev; + } + + mdev_devres = devres_alloc(media_device_release_devres, + sizeof(struct media_device_devres), + GFP_KERNEL); + if (!mdev_devres) return NULL; - return devres_get(dev, mdev, NULL, NULL); + + ptr = devres_get(dev, mdev_devres, NULL, NULL); + if (ptr) + kref_init(&ptr->kref); + else + devres_free(mdev_devres); + + return &ptr->mdev; } EXPORT_SYMBOL_GPL(media_device_get_devres); struct media_device *media_device_find_devres(struct device *dev) { - return devres_find(dev, media_device_release_devres, NULL, NULL); + struct media_device_devres *mdev_devres; + + mdev_devres = devres_find(dev, media_device_release_devres, NULL, NULL); + if (!mdev_devres) + return NULL; + + return &mdev_devres->mdev; } EXPORT_SYMBOL_GPL(media_device_find_devres); diff --git a/drivers/media/usb/au0828/au0828-core.c b/drivers/media/usb/au0828/au0828-core.c index 06da73f1ff22..060904ed8f20 100644 --- a/drivers/media/usb/au0828/au0828-core.c +++ b/drivers/media/usb/au0828/au0828-core.c @@ -157,8 +157,7 @@ static void au0828_unregister_media_device(struct au0828_dev *dev) dev->media_dev->enable_source = NULL; dev->media_dev->disable_source = NULL; - media_device_unregister(dev->media_dev); - media_device_cleanup(dev->media_dev); + media_device_unregister_devres(dev->media_dev); dev->media_dev = NULL; #endif } diff --git a/include/media/media-device.h b/include/media/media-device.h index b21ef244ad3e..e59772ed8494 100644 --- a/include/media/media-device.h +++ b/include/media/media-device.h @@ -23,6 +23,7 @@ #ifndef _MEDIA_DEVICE_H #define _MEDIA_DEVICE_H +#include <linux/kref.h> #include <linux/list.h> #include <linux/mutex.h> @@ -382,6 +383,16 @@ struct media_device { unsigned int notification); }; +/** + * struct media_device_devres - Media device device resource + * @mdev: pointer to struct media_device + * @kref: Object refcount + */ +struct media_device_devres { + struct media_device mdev; + struct kref kref; +}; + /* We don't need to include pci.h or usb.h here */ struct pci_dev; struct usb_device; @@ -604,6 +615,19 @@ struct media_device *media_device_get_devres(struct device *dev); */ struct media_device *media_device_find_devres(struct device *dev); +/** + * media_device_unregister_devres) - Unregister media device allocated as + * as device resource + * + * @dev: pointer to struct &device. + * + * Devices allocated via media_device_get_devres should be de-alocalted + * and freed via this function. Callers should not call + * media_device_unregister() nor media_device_cleanup() on devices + * allocated via media_device_get_devres(). + */ +void media_device_unregister_devres(struct media_device *mdev); + /* Iterate over all entities. */ #define media_device_for_each_entity(entity, mdev) \ list_for_each_entry(entity, &(mdev)->entities, graph_obj.list) @@ -688,6 +712,10 @@ static inline struct media_device *media_device_find_devres(struct device *dev) return NULL; } +static inline void media_device_unregister_devres(struct media_device *mdev) +{ +} + static inline void media_device_pci_init(struct media_device *mdev, struct pci_dev *pci_dev, char *name) diff --git a/sound/usb/media.c b/sound/usb/media.c index 93a50d01490c..f78955fd0d6e 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -311,8 +311,7 @@ void media_snd_device_delete(struct snd_usb_audio *chip) media_snd_mixer_delete(chip); if (mdev) { - if (media_devnode_is_registered(&mdev->devnode)) - media_device_unregister(mdev); + media_device_unregister_devres(mdev); chip->media_dev = NULL; } } -- 2.5.0

3 9

[alsa-devel] [PATCH] ucm: add cset-tlv command
by Hsin-Yu Chao 23 Mar '16

23 Mar '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 80 ++++++++++++++++++++++++++++++++++++++++++++--------- src/ucm/parser.c | 10 +++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 78 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..a4ccb65 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,47 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int tlv_parse(char **res, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + free(*res); + *res = NULL; + } + + __fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +267,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + char *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +283,32 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + err = tlv_parse(&res, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, (unsigned int *)res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +317,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +351,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.1.2

2 5

[alsa-devel] [PATCH V3 0/6] alsabat: clean structure and tinyalsa support
by han.lu＠intel.com 23 Mar '16

23 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> 1. Refactoring alsa process to make structure simpler and cleaner, use general functions and unified return value for convenience of maintaining; 2. Add tinyalsa support, user can choose tinyalsa with "configure --enable-alsabat-backend-tiny" option. By default ALSA lib is used. The intention is for alsabat to run on tinyalsa platforms such as Android and some IoT devices. changes on v3: 1. use option "--enable-alsabat-backend-tiny" instead of "--enable-tinyalsa" to avoid ambiguous. 2. fix inconsistent in man page and configure.ac. changes on v2: 1. add more description to explain the purpose of each patch. 2. use configure option to select tinyalsa, rather than auto detect environment and decide. Lu, Han (6): alsabat: refactoring alsa capture thread alsabat: use general function for wav header update alsabat: clean return value for playback and capture threads alsabat: use general data generator function alsabat: move alsa process to a single block alsabat: add tinyalsa support bat/Makefile.am | 12 +- bat/alsa.c | 217 +++++++++++--------------- bat/alsabat.1 | 3 + bat/bat.c | 58 +++---- bat/common.c | 68 +++++++++ bat/common.h | 17 ++- bat/signal.c | 13 +- bat/tinyalsa.c | 460 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 20 +++ configure.ac | 17 +++ 10 files changed, 711 insertions(+), 174 deletions(-) create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

2 7

[alsa-devel] [PATCH 1/3] ALSA: hda - Asus N750JV external subwoofer fixup
by Takashi Iwai 23 Mar '16

23 Mar '16

From: Bobi Mihalca <bobbymihalca(a)touchtech.ro> ASUS N750JV needs the same fixup as N550 for enabling its subwoofer. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115181 Signed-off-by: Bobi Mihalca <bobbymihalca(a)touchtech.ro> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 4f5ca0b9ce27..44c52c2b7be7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6670,6 +6670,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), + SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), -- 2.7.4

1 2

[alsa-devel] [PATCH] ASoC: rt5645: Add dmi_system_id "Google Setzer"
by John Lin 23 Mar '16

23 Mar '16

Add platform specific data for Setzer project. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 93e8c90..75e5a4a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3552,6 +3552,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Setzer", + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ucm: add cset-tlv
by Hsin-Yu Chao 23 Mar '16

23 Mar '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 91 +++++++++++++++++++++++++++++++++++++++++++++-------- src/ucm/parser.c | 10 ++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 89 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..c6bbf27 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,54 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int read_tlv_file(char **res, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + unsigned int *tlv; + + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + if (sz > 1024 * 1024 || sz < 8) { + uc_error("unsupported file size"); + goto __fail; + } + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + free(*res); + *res = NULL; + } + /* Check if the tlv file specifies valid size. */ + tlv = (unsigned int *)(*res); + if (tlv[1] + 2 * sizeof(unsigned int) != sz) { + uc_error("Invalid tlv size"); + free(*res); + *res = NULL; + } + + __fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +274,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + char *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +290,36 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + goto __fail; + } + err = read_tlv_file(&res, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, (unsigned int *)res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +328,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +362,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.1.2

1 0

[alsa-devel] [PATCH v3] ALSA: timer: fix gparams ioctl compatibility for different architectures
by Takashi Sakamoto 23 Mar '16

23 Mar '16

'struct snd_timer_gparams' includes some members with 'unsigned long', therefore its size differs depending on data models of architecture. As a result, x86/x32 applications fail to execute ioctl(2) with SNDRV_TIMER_GPARAMS command on x86_64 machine. This commit fixes this bug by adding a pair of structure and ioctl command for the compatibility. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/timer.c | 20 +++++++++++++------- sound/core/timer_compat.c | 30 +++++++++++++++++++++++++++++- 2 files changed, 42 insertions(+), 8 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index aa1b15c..ea4d999 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1502,17 +1502,13 @@ static int snd_timer_user_ginfo(struct file *file, return err; } -static int snd_timer_user_gparams(struct file *file, - struct snd_timer_gparams __user *_gparams) +static int timer_set_gparams(struct snd_timer_gparams *gparams) { - struct snd_timer_gparams gparams; struct snd_timer *t; int err; - if (copy_from_user(&gparams, _gparams, sizeof(gparams))) - return -EFAULT; mutex_lock(&register_mutex); - t = snd_timer_find(&gparams.tid); + t = snd_timer_find(&gparams->tid); if (!t) { err = -ENODEV; goto _error; @@ -1525,12 +1521,22 @@ static int snd_timer_user_gparams(struct file *file, err = -ENOSYS; goto _error; } - err = t->hw.set_period(t, gparams.period_num, gparams.period_den); + err = t->hw.set_period(t, gparams->period_num, gparams->period_den); _error: mutex_unlock(&register_mutex); return err; } +static int snd_timer_user_gparams(struct file *file, + struct snd_timer_gparams __user *_gparams) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams, _gparams, sizeof(gparams))) + return -EFAULT; + return timer_set_gparams(&gparams); +} + static int snd_timer_user_gstatus(struct file *file, struct snd_timer_gstatus __user *_gstatus) { diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 2e90822..6a437eb 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -22,6 +22,19 @@ #include <linux/compat.h> +/* + * ILP32/LP64 has different size for 'long' type. Additionally, the size + * of storage alignment differs depending on architectures. Here, '__packed' + * qualifier is used so that the size of this structure is multiple of 4 and + * it fits to any architectures with 32 bit storage alignment. + */ +struct snd_timer_gparams32 { + struct snd_timer_id tid; + u32 period_num; + u32 period_den; + unsigned char reserved[32]; +} __packed; + struct snd_timer_info32 { u32 flags; s32 card; @@ -32,6 +45,19 @@ struct snd_timer_info32 { unsigned char reserved[64]; }; +static int snd_timer_user_gparams_compat(struct file *file, + struct snd_timer_gparams32 __user *user) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams.tid, &user->tid, sizeof(gparams.tid)) || + get_user(gparams.period_num, &user->period_num) || + get_user(gparams.period_den, &user->period_den)) + return -EFAULT; + + return timer_set_gparams(&gparams); +} + static int snd_timer_user_info_compat(struct file *file, struct snd_timer_info32 __user *_info) { @@ -99,6 +125,7 @@ static int snd_timer_user_status_compat(struct file *file, */ enum { + SNDRV_TIMER_IOCTL_GPARAMS32 = _IOW('T', 0x04, struct snd_timer_gparams32), SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32), SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32), #ifdef CONFIG_X86_X32 @@ -114,7 +141,6 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PVERSION: case SNDRV_TIMER_IOCTL_TREAD: case SNDRV_TIMER_IOCTL_GINFO: - case SNDRV_TIMER_IOCTL_GPARAMS: case SNDRV_TIMER_IOCTL_GSTATUS: case SNDRV_TIMER_IOCTL_SELECT: case SNDRV_TIMER_IOCTL_PARAMS: @@ -128,6 +154,8 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: return snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + case SNDRV_TIMER_IOCTL_GPARAMS32: + return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: return snd_timer_user_info_compat(file, argp); case SNDRV_TIMER_IOCTL_STATUS32: -- 2.7.3

2 1

[alsa-devel] [PATCH] ALSA: timer: fix gparams ioctl compatibility for different architectures
by Takashi Sakamoto 23 Mar '16

23 Mar '16

'struct snd_timer_gparams' includes some members with 'unsigned long', therefore its size differs depending on data models of architecture. As a result, x86/x32 applications fail to execute ioctl(2) with SNDRV_TIMER_GPARAMS command on x86_64 machine. This commit fixes this bug by adding a pair of structure and ioctl command for the compatibility. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/timer.c | 20 +++++++++++++------- sound/core/timer_compat.c | 34 +++++++++++++++++++++++++++++++++- 2 files changed, 46 insertions(+), 8 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index aa1b15c..ea4d999 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1502,17 +1502,13 @@ static int snd_timer_user_ginfo(struct file *file, return err; } -static int snd_timer_user_gparams(struct file *file, - struct snd_timer_gparams __user *_gparams) +static int timer_set_gparams(struct snd_timer_gparams *gparams) { - struct snd_timer_gparams gparams; struct snd_timer *t; int err; - if (copy_from_user(&gparams, _gparams, sizeof(gparams))) - return -EFAULT; mutex_lock(&register_mutex); - t = snd_timer_find(&gparams.tid); + t = snd_timer_find(&gparams->tid); if (!t) { err = -ENODEV; goto _error; @@ -1525,12 +1521,22 @@ static int snd_timer_user_gparams(struct file *file, err = -ENOSYS; goto _error; } - err = t->hw.set_period(t, gparams.period_num, gparams.period_den); + err = t->hw.set_period(t, gparams->period_num, gparams->period_den); _error: mutex_unlock(&register_mutex); return err; } +static int snd_timer_user_gparams(struct file *file, + struct snd_timer_gparams __user *_gparams) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams, _gparams, sizeof(gparams))) + return -EFAULT; + return timer_set_gparams(&gparams); +} + static int snd_timer_user_gstatus(struct file *file, struct snd_timer_gstatus __user *_gstatus) { diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 2e90822..d10c792 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -22,6 +22,19 @@ #include <linux/compat.h> +/* + * ILP32/LP64 has different size for 'long' type. Additionally, the size + * of storage alignment differs depending on architectures. Here, '__packed' + * qualifier is used so that the size of this structure is multiple of 4 and + * it fits to any architectures with 32 bit storage alignment. + */ +struct snd_timer_gparams32 { + struct snd_timer_id tid; + u32 period_num; + u32 period_den; + unsigned char reserved[32]; +} __packed; + struct snd_timer_info32 { u32 flags; s32 card; @@ -32,6 +45,23 @@ struct snd_timer_info32 { unsigned char reserved[64]; }; +static int snd_timer_user_gparams_compat(struct file *file, + struct snd_timer_gparams32 __user *user) +{ + struct snd_timer_gparams gparams; + + if (get_user(gparams.tid.dev_class, &user->tid.dev_class) || + get_user(gparams.tid.dev_sclass, &user->tid.dev_sclass) || + get_user(gparams.tid.card, &user->tid.card) || + get_user(gparams.tid.device, &user->tid.device) || + get_user(gparams.tid.subdevice, &user->tid.subdevice) || + get_user(gparams.period_num, &user->period_num) || + get_user(gparams.period_den, &user->period_den)) + return -EFAULT; + + return timer_set_gparams(&gparams); +} + static int snd_timer_user_info_compat(struct file *file, struct snd_timer_info32 __user *_info) { @@ -99,6 +129,7 @@ static int snd_timer_user_status_compat(struct file *file, */ enum { + SNDRV_TIMER_IOCTL_GPARAMS32 = _IOW('T', 0x04, struct snd_timer_gparams32), SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32), SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32), #ifdef CONFIG_X86_X32 @@ -114,7 +145,6 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PVERSION: case SNDRV_TIMER_IOCTL_TREAD: case SNDRV_TIMER_IOCTL_GINFO: - case SNDRV_TIMER_IOCTL_GPARAMS: case SNDRV_TIMER_IOCTL_GSTATUS: case SNDRV_TIMER_IOCTL_SELECT: case SNDRV_TIMER_IOCTL_PARAMS: @@ -128,6 +158,8 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: return snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + case SNDRV_TIMER_IOCTL_GPARAMS32: + return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: return snd_timer_user_info_compat(file, argp); case SNDRV_TIMER_IOCTL_STATUS32: -- 2.7.3

2 2

[alsa-devel] [PATCH 0/3] ARM: OMAP3: Fix McBSP2/3 hwmod setup for sidetone
by Peter Ujfalusi 22 Mar '16

22 Mar '16

Hi, The series addresses a long standing issue with McBSP2/3 regarding to hwmod setup. When booting with DT a warning is printed that mcbsp2/3 is using two hwmod. The root of the issue is the way how the hwmod data was constructed in the first place for OMAP3 McBSP2/3. After re-reading the TRM it is clear that the sidetone should not have it's own hwmod data as it is not a separate IP, it is part of the McBSP module. It can not affect PRCM either since it's SYSCONFIG register's AUTOIDLE bit is only sets the autoidle from the internal McBSP_iclk clock to the sidetone block of the same McBSP. The first two patch will remove the omap2/3_sidetone hwmod and it's use from DT. The ASoC patch is clear the autoidle bit in McBSP sidetone as it is adviced in the TRM. Mark: The ASoC patch can be safely merged via ASoC tree. Regards, Peter --- Peter Ujfalusi (3): ARM: DTS: omap3: Remove mcbsp2/3_sidetone hwmod reference for McBSP2/3 ARM: OMAP3: hwmod data: Merge and remove the McBSP sidetone related data ASoC: omap-mcbsp: Enable/disable sidetone block auto clock gating for omap3 arch/arm/boot/dts/omap3.dtsi | 4 +- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 120 ++++------------------------- sound/soc/omap/mcbsp.c | 8 ++ 3 files changed, 24 insertions(+), 108 deletions(-) -- 2.7.3

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[alsa-devel] [PATCH] sound/usb: fix to release stream resources from media_snd_device_delete()
by Shuah Khan 22 Mar '16

22 Mar '16

Fix to release stream resources from media_snd_device_delete() before media device is unregistered. Without this change, stream resource free is attempted after the media device is unregistered which would result in use-after-free errors. Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- - Ran bind/unbind loop (1000 iteration) test on snd-usb-audio while running mc_nextgen_test loop (1000 iterations) in parallel. - Ran bind/unbind and rmmod/modprobe tests on both drivers. Also generated graphs when after bind/unbind, rmmod/modprobe. Graphs look good. - Note: Please apply the following patch to fix memory leak: sound/usb: Fix memory leak in media_snd_stream_delete() during unbind https://lkml.org/lkml/2016/3/16/1050 sound/usb/media.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/media.c b/sound/usb/media.c index de4a815..e35af88 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -301,6 +301,13 @@ int media_snd_device_create(struct snd_usb_audio *chip, void media_snd_device_delete(struct snd_usb_audio *chip) { struct media_device *mdev = chip->media_dev; + struct snd_usb_stream *stream; + + /* release resources */ + list_for_each_entry(stream, &chip->pcm_list, list) { + media_snd_stream_delete(&stream->substream[0]); + media_snd_stream_delete(&stream->substream[1]); + } media_snd_mixer_delete(chip); -- 2.5.0

2 9

[alsa-devel] [PATCH V2 0/6] alsabat: clean structure and tinyalsa support
by han.lu＠intel.com 22 Mar '16

22 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> 1. refactoring alsa process to make structure simpler and cleaner, use general functions and unified return value for convenience of maintaining; 2. add tinyalsa support, user can choose tinyalsa with "configure --enable-tinyalsa" option. By default ALSA lib is used. The purpose is for alsabat to run on tinyalsa platforms such as Android and some IoT devices. Lu, Han (6): alsabat: refactoring alsa capture thread alsabat: use general function for wav header update alsabat: clean return value for playback and capture threads alsabat: use general data generator function alsabat: move alsa process to a single block alsabat: add tinyalsa support bat/Makefile.am | 12 +- bat/alsa.c | 217 +++++++++++--------------- bat/alsabat.1 | 3 + bat/bat.c | 58 +++---- bat/common.c | 68 +++++++++ bat/common.h | 17 ++- bat/signal.c | 13 +- bat/tinyalsa.c | 460 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 20 +++ configure.ac | 17 +++ 10 files changed, 711 insertions(+), 174 deletions(-) create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

2 7

[alsa-devel] [PATCH 0/6] *** alsabat: clean structure and tinyalsa support ***
by han.lu＠intel.com 22 Mar '16

22 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> 1. Clean structure, use general function to replace local processes. 2. Add tinyalsa support, configure will read environment and decide to link to ALSA lib or tinyalsa lib. Lu, Han (6): alsabat: clean file process on capture thread loop alsabat: use common wav process function in playback loop alsabat: clean return value of playback and capture loops alsabat: use common data generator function alsabat: move alsa process to alsa.c alsabat: add tinyalsa support bat/Makefile.am | 12 +- bat/alsa.c | 217 +++++++++++--------------- bat/alsabat.1 | 3 + bat/bat.c | 58 +++---- bat/common.c | 68 +++++++++ bat/common.h | 17 ++- bat/signal.c | 13 +- bat/tinyalsa.c | 460 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 20 +++ configure.ac | 4 + 10 files changed, 698 insertions(+), 174 deletions(-) create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

3 10

[alsa-devel] [PATCH] ALSA: timer: fix another ioctl compatibility bug
by Takashi Sakamoto 22 Mar '16

22 Mar '16

Hi, In ALSA timer interface, I found another compatibility bug to different ABIs. This patch is to fix it. I note that the bug is in really, really-minor functionality and no drivers use it, therefore it has little impact to this subsystem. This patch is just for manners between kernel/userspace interfaces. To check the compatibility, I wrote a series of easy and small programs for each ALSA interface. It's available in my repository on github. https://github.com/takaswie/alsa-ioctl-test As long as I tested with these programs, this patch is the last fix for the compatibility issues which comes from differences of data model. On the other hand, I confirmed that each compatibility layer is not fully compatible to core implementation. For example, ioctl() with SNDRV_PCM_IOCTL_WRITEI_FRAMES and argument filled by zero for PCM capture character device (not for playback device) returns different error codes for each of x86/x86_64 binary. I think this should be fixed, but don't start to work for it yet. Regards Takashi Sakamoto (1): ALSA: timer: fix ioctl compatibility for different data models sound/core/timer_compat.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) -- 2.7.3

2 3

[alsa-devel] [PATCH 0/7] Intel HD-audio HDMI codec driver rewrites
by Takashi Iwai 22 Mar '16

22 Mar '16

Hi, here is a patch series for rewrite / cleanup of Intel HD-audio codec driver. The main purpose of this patchset is to re-add the support of i915 ELD notification for old chipsets. For that, it starts from the cleanup of the HDMI codec driver code to get rid of excessive usages of is_*() macros. Then the fixes for supporting the old chips in HDA core side and HDMI codec driver side. Although this could be included for 4.6, my current plan is to postpone this for 4.7. Takashi === Takashi Iwai (7): ALSA: hda - Split out Intel-specific codes from patch_generic_hdmi() ALSA: hda - Apply AMP fix in hdmi_setup_audio_infoframe() generically ALSA: hda - Override HDMI setup_stream ops for Intel HSW+ ALSA: hda - Introduce pin_cvt_fixup() ops to hdmi parser ALSA: hda - Use eld notifier for Intel SandyBridge and IvyBridge HDMI/DP ALSA: hda - Add the pin / port mapping on Intel ILK and VLV ALSA: hda - Enable i915 ELD notifier for Intel IronLake and Baytrail include/sound/hda_i915.h | 10 +- sound/hda/hdac_i915.c | 45 ++++-- sound/pci/hda/patch_hdmi.c | 373 ++++++++++++++++++++++++++++++--------------- 3 files changed, 293 insertions(+), 135 deletions(-) -- 2.7.4

1 7

[alsa-devel] [PATCH 0/3] *** Some trival fixes on alsabat ***
by han.lu＠intel.com 21 Mar '16

21 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Fix misusing of errno, fopen, prints and return value Lu, Han (3): alsabat: fix misusing of errno alsabat: fix fopen and messages alsabat: use variable for thread return value bat/alsa.c | 50 ++++++++++++++++++++++++++------------------------ bat/analyze.c | 6 +++--- bat/bat.c | 39 ++++++++++++++++++++++----------------- bat/common.c | 1 - 4 files changed, 51 insertions(+), 45 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 21 Mar '16

21 Mar '16

Driver for TI's TAS5720L/M digital audio amplifiers. The driver should be pretty standard except the optional interrupt-based fault reporting. Some background on the fault reporting since that might be a discussion of interest. The code should have that documented rather well but I wanted to bring it up right away. The TAS5720 devices can report conditions such as over-current (short), over temp, and others that can be important to know in a real-world system. However the device lacks a way to mask the generation of fault interrupts except by shutting it down, which can lead to cases where the device generates SAIF-fault host interrupts every 300us! This is clearly a lot of overhead to be thrown at the SoC (and my bench testing has demonstrated that) so after trying different things to tame this behavior I ended up implementing method to disable/enable the interrupts in the driver itself in combination with rate-limiting the processing of those events. The fact of processing interrupts in a threaded fashion (IRQF_ONESHOT) alone was not enough under certain conditions to prevent the SoC from not being able to get to all interrupts even with an empty threaded handler, and it would freeze and eventually the Kernel would disable the interrupt ("no one cared"). The scheme currently implemented seems to work reliably cycling different audio files over hours also forcing error conditions. But if anybody has a "cleaner" idea how to rate-limit IRQs on a very low level (currently done in the IRQ handler function) I'm absolutely open to suggestions. This all may not be an issue at all when using a faster/lower-latency SoC than what I'm using (BeagleBone Black) but the fact that IRQF_ONESHOT alone was not enough to prevent the system to be brought to its knees (again, even with an empty threaded handler) was a bit of a surprise to me. -- Andreas Dannenberg Texas Instruments Inc Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 37 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 733 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 872 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

2 4

[alsa-devel] New snd-hda warning spew
by Ville Syrjälä 21 Mar '16

21 Mar '16

We have a few new WARN spews from snd-hda causing some grief in i915 CI. This one happens on ILK and BYT. Looks like it happens 100% of the time on driver load: [ 18.809850] ------------[ cut here ]------------ [ 18.809866] WARNING: CPU: 0 PID: 39 at sound/hda/hdac_i915.c:129 pin2port+0x25/0x30 [snd_hda_core]() [ 18.809869] Modules linked in: ax88179_178a usbnet mii intel_powerclamp coretemp crct10dif_pclmul i915 snd_hda_codec_hdmi snd_hda_codec_generic crc32_pclmul ghash_clmulni_intel snd_hda_intel snd_hda_codec snd_hwdep snd_hda_core snd_pcm mei_me mei lpc_ich sdhci_pci sdhci mmc_core e1000e ptp pps_core [ 18.809903] CPU: 0 PID: 39 Comm: kworker/0:1 Not tainted 4.5.0-gfxbench+ #1 [ 18.809905] Hardware name: Hewlett-Packard HP EliteBook 8440p/172A, BIOS 68CCU Ver. F.24 09/13/2013 [ 18.809912] Workqueue: events azx_probe_work [snd_hda_intel] [ 18.809915] 0000000000000000 ffff8800b646b9e8 ffffffff813fef15 0000000000000000 [ 18.809920] ffffffffa00f277c ffff8800b646ba20 ffffffff81078a21 ffff8800b2436000 [ 18.809924] ffff8800b24755e8 ffff8800b24755f0 0000000000000100 ffff8800aed08330 [ 18.809929] Call Trace: [ 18.809936] [<ffffffff813fef15>] dump_stack+0x67/0x92 [ 18.809942] [<ffffffff81078a21>] warn_slowpath_common+0x81/0xc0 [ 18.809945] [<ffffffff81078b15>] warn_slowpath_null+0x15/0x20 [ 18.809953] [<ffffffffa00f12d5>] pin2port+0x25/0x30 [snd_hda_core] [ 18.809960] [<ffffffffa00f1378>] snd_hdac_acomp_get_eld+0x38/0x70 [snd_hda_core] [ 18.809966] [<ffffffffa017da52>] hdmi_present_sense+0x92/0x370 [snd_hda_codec_hdmi] [ 18.809971] [<ffffffffa017e0ff>] generic_hdmi_build_controls+0x11f/0x200 [snd_hda_codec_hdmi] [ 18.809985] [<ffffffffa01105b1>] snd_hda_codec_build_controls+0x191/0x1d0 [snd_hda_codec] [ 18.809993] [<ffffffffa0110881>] ? snd_hda_codec_build_pcms+0xe1/0x1a0 [snd_hda_codec] [ 18.809999] [<ffffffffa010b442>] hda_codec_driver_probe+0x82/0x100 [snd_hda_codec] [ 18.810005] [<ffffffff8153d729>] driver_probe_device+0x229/0x450 [ 18.810008] [<ffffffff8153da7d>] __device_attach_driver+0x6d/0x80 [ 18.810011] [<ffffffff8153da10>] ? driver_allows_async_probing+0x30/0x30 [ 18.810015] [<ffffffff8153b4b8>] bus_for_each_drv+0x58/0x90 [ 18.810018] [<ffffffff8153d3c8>] __device_attach+0xb8/0x130 [ 18.810022] [<ffffffff8153dc0e>] device_initial_probe+0xe/0x10 [ 18.810025] [<ffffffff8153c76e>] bus_probe_device+0x9e/0xb0 [ 18.810028] [<ffffffff8153a5a1>] device_add+0x421/0x5f0 [ 18.810035] [<ffffffff817c2599>] ? mutex_unlock+0x9/0x10 [ 18.810041] [<ffffffffa00eba7f>] snd_hdac_device_register+0xf/0x40 [snd_hda_core] [ 18.810048] [<ffffffffa010b1f6>] snd_hda_codec_configure+0x36/0x180 [snd_hda_codec] [ 18.810057] [<ffffffffa01150c3>] azx_codec_configure+0x23/0x40 [snd_hda_codec] [ 18.810062] [<ffffffffa0134707>] azx_probe_work+0x567/0x870 [snd_hda_intel] [ 18.810067] [<ffffffff81093c8b>] process_one_work+0x1cb/0x680 [ 18.810070] [<ffffffff81093c06>] ? process_one_work+0x146/0x680 [ 18.810073] [<ffffffff81094189>] worker_thread+0x49/0x490 [ 18.810076] [<ffffffff81094140>] ? process_one_work+0x680/0x680 [ 18.810081] [<ffffffff8109a87a>] kthread+0xea/0x100 [ 18.810085] [<ffffffff817c5407>] ? _raw_spin_unlock_irq+0x27/0x50 [ 18.810090] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 18.810093] [<ffffffff817c60bf>] ret_from_fork+0x3f/0x70 [ 18.810097] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 18.810277] ---[ end trace 2d4983bcfed2cd4b ]--- This other one was seen at least on on SKL: [ 124.808525] ------------[ cut here ]------------ [ 124.808545] WARNING: CPU: 3 PID: 173 at sound/hda/hdac_i915.c:91 snd_hdac_display_power+0xf1/0x110 [snd_hda_core]() [ 124.808551] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic x86_pkg_temp_thermal intel_powerclamp coretemp i915 crct10dif_pclmul snd_hda_intel crc32_pclmul snd_hda_codec ghash_clmulni_intel snd_hwdep snd_hda_core snd_pcm mei_me mei i2c_hid sdhci_pci e1000e ptp sdhci pps_core mmc_core [ 124.808603] CPU: 3 PID: 173 Comm: kworker/3:2 Tainted: G W 4.5.0-gfxbench+ #1 [ 124.808608] Hardware name: /NUC6i5SYB, BIOS SYSKLi35.86A.0028.2015.1112.1822 11/12/2015 [ 124.808616] Workqueue: pm pm_runtime_work [ 124.808639] 0000000000000000 ffff88026423fbb8 ffffffff813fef15 0000000000000000 [ 124.808644] ffffffffa00f877c ffff88026423fbf0 ffffffff81078a21 0000000000000000 [ 124.808648] ffff880085a11148 ffff880085586ab8 ffff8800861b8000 0000000000000000 [ 124.808652] Call Trace: [ 124.808656] [<ffffffff813fef15>] dump_stack+0x67/0x92 [ 124.808660] [<ffffffff81078a21>] warn_slowpath_common+0x81/0xc0 [ 124.808662] [<ffffffff81078b15>] warn_slowpath_null+0x15/0x20 [ 124.808668] [<ffffffffa00f77e1>] snd_hdac_display_power+0xf1/0x110 [snd_hda_core] [ 124.808673] [<ffffffffa015039d>] azx_intel_link_power+0xd/0x10 [snd_hda_intel] [ 124.808680] [<ffffffffa011e32a>] azx_link_power+0x1a/0x30 [snd_hda_codec] [ 124.808684] [<ffffffffa00f21f9>] snd_hdac_link_power+0x29/0x40 [snd_hda_core] [ 124.808690] [<ffffffffa01192a6>] hda_codec_runtime_suspend+0x76/0xa0 [snd_hda_codec] [ 124.808697] [<ffffffffa0119230>] ? hda_codec_runtime_resume+0x50/0x50 [snd_hda_codec] [ 124.808701] [<ffffffff8154719d>] __rpm_callback+0x2d/0x70 [ 124.808704] [<ffffffff815471ff>] rpm_callback+0x1f/0x80 [ 124.808710] [<ffffffffa0119230>] ? hda_codec_runtime_resume+0x50/0x50 [snd_hda_codec] [ 124.808713] [<ffffffff81547764>] rpm_suspend+0x134/0x7f0 [ 124.808716] [<ffffffff810c80a9>] ? __lock_is_held+0x49/0x70 [ 124.808719] [<ffffffff81548129>] rpm_idle+0x209/0x530 [ 124.808722] [<ffffffff815494f9>] pm_runtime_work+0xa9/0xc0 [ 124.808725] [<ffffffff81093c8b>] process_one_work+0x1cb/0x680 [ 124.808728] [<ffffffff81093c06>] ? process_one_work+0x146/0x680 [ 124.808731] [<ffffffff81094189>] worker_thread+0x49/0x490 [ 124.808733] [<ffffffff81094140>] ? process_one_work+0x680/0x680 [ 124.808735] [<ffffffff81094140>] ? process_one_work+0x680/0x680 [ 124.808738] [<ffffffff8109a87a>] kthread+0xea/0x100 [ 124.808742] [<ffffffff817c53d7>] ? _raw_spin_unlock_irq+0x27/0x50 [ 124.808745] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 124.808749] [<ffffffff817c607f>] ret_from_fork+0x3f/0x70 [ 124.808752] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 124.808755] ---[ end trace 2b97e417eb08d213 ]--- -- Ville Syrjälä Intel OTC

4 23

[alsa-devel] amidi.c.patch review request
by Martin Tarenskeen 21 Mar '16

21 Mar '16

Hi, I improved my own patch to amidi.c As a result I can now succesfully fetch a SysEx dump from my Yamaha Reface DX, and it exits succesfully after the timeout that I specify using the -t option. Problem: My synth sends continuously not only 0xFE (Active Sensing) but also 0xF8 (Timing Clock). 24 times per quarter note to be exactly. With a fast tempo like 250 BPM this gives a 0xF8 byte every 10 ms. Until now when I tried to fetch a SysEx dump from my synth using amidi the data were polluted with lots of 0xF8 bytes and the -t option would not work. A bug I would say. The attached patch seems to solve this issue, but please comment and improve where needed. Please try with other synths also. Can someone submit my patch to the official sources? I don't know how to. See attached. -- Martin

2 1

Re: [alsa-devel] [Alsa-devel] [PATCH] amidi active sensing option
by Clemens Ladisch 21 Mar '16

21 Mar '16

Martin Tarenskeen wrote: > I searched on Google and found in the old alsa-devel archives the > anouncement of this option to enable/disable detecting Active Sensing > (0xfe) bytes with the amidi utility. > But not only 0xfe (Active Sensing) should be blocked by default, also > other Midi System Realtime messages. In other SysEx dump utilities > that I know there is a simialr option called something like > "rt-filter" or "realtime-filter". > [...] > I have attached my patch. Thank you for this improvement. However, the name of the option ("--active-sensing") is no longer correct, and the man page must be updated, too. Also, patches need a Signed-off-by line to indicate that the submitter accepts the DCO; see <http://elinux.org/Developer_Certificate_Of_Origin>. Regards, Clemens

1 0

[alsa-devel] [PATCH] sound: usb: add Microsoft HD-5001 to quirks
by Victor Clément 20 Mar '16

20 Mar '16

The Microsoft HD-5001 webcam microphone does not support sample rate reading as the HD-5000 one. This results in dmesg errors and sound hanging with pulseaudio. Signed-off-by: Victor Clément <victor.clement(a)openmailbox.org> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4f6ce1c..f27016f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1121,6 +1121,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) switch (chip->usb_id) { case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ + case USB_ID(0x045E, 0x076E): /* MS Lifecam HD-5001 */ case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ -- 2.7.2

2 1

[alsa-devel] [GIT PULL]: Update sst audio firmware for Skylake
by Vinod Koul 19 Mar '16

19 Mar '16

Hello Linux firmware maintainers, Please PULL to receive the updated version of Skylake audio firmware version 896. This contains some bug fixes and wov module fixes. The following changes since commit 9be9ff282635c25694819c39aae0f6cc6c251c12: iwlwifi: add -21.ucode to WHENCE (2016-03-10 11:28:28 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to 780f4eed8ff971bd4d6c41ecded4c99135cd36f9: linux-firmware: intel: Update Skylake audio firmware (2016-03-14 11:38:19 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v896.bin | Bin 0 -> 233472 bytes 2 files changed, 1 insertion(+), 1 deletion(-) create mode 100644 intel/dsp_fw_release_v896.bin Thanks -- ~Vinod

2 1

[alsa-devel] USB Sound oops in Linus's latest tree
by Greg KH 19 Mar '16

19 Mar '16

Hi, I'm getting the following oops with my USB sound device using Linus's latest tree of the moment, which has the sound tree pull in it. Anyone seen this before? thanks, greg k-h [ +0.002298] input: Schiit Audio USB Modi Device as /devices/pci0000:00/0000:00:14.0/usb2/2-10/2-10.1/2-10.1:1.2/0003:0D8C:1319.0001/input/input17 [ +0.000056] hid-generic 0003:0D8C:1319.0001: input,hidraw0: USB HID v1.00 Device [Schiit Audio USB Modi Device] on usb-0000:00:14.0-10.1/input2 [ +0.008749] BUG: unable to handle kernel NULL pointer dereference at 0000000000000014 [ +0.001151] IP: [<ffffffffc0e79921>] usb_audio_probe+0x1e1/0x8f0 [snd_usb_audio] [ +0.001142] PGD 0 [ +0.001114] Oops: 0000 [#1] PREEMPT SMP [ +0.001115] Modules linked in: snd_usb_audio(+) hid_generic snd_usbmidi_lib usbhid snd_rawmidi snd_seq_device hid media arc4 fuse btusb btrtl btbcm btintel bluetooth rt2800usb rt2x00usb rt2800lib rt2x00lib mac80211 crc_ccitt cfg80211 rfkill nls_cp437 vfat fat iTCO_wdt iTCO_vendor_support mxm_wmi x86_pkg_temp_thermal intel_powerclamp coretemp kvm_intel kvm irqbypass crct10dif_pclmul crct10dif_common crc32_pclmul crc32c_intel ghash_clmulni_intel aesni_intel aes_x86_64 glue_helper lrw gf128mul ablk_helper cryptd psmouse serio_raw pcspkr firewire_ohci firewire_core crc_itu_t xhci_pci ehci_pci xhci_hcd ehci_hcd mei_me usbcore mei lpc_ich mfd_core usb_common wmi evdev ip_tables x_tables [ +0.004079] CPU: 4 PID: 615 Comm: systemd-udevd Not tainted 4.6.0-rc1+ #103 [ +0.001382] Hardware name: / , BIOS KLZ8711D.86A.0457.2015.0713.1319 07/13/2015 [ +0.001387] task: ffff880098ec0e00 ti: ffff880251920000 task.ti: ffff880251920000 [ +0.001417] RIP: 0010:[<ffffffffc0e79921>] [<ffffffffc0e79921>] usb_audio_probe+0x1e1/0x8f0 [snd_usb_audio] [ +0.001414] RSP: 0018:ffff880251923af0 EFLAGS: 00010246 [ +0.001409] RAX: 0000000000000000 RBX: ffff88024d602000 RCX: 0000000000000000 [ +0.001404] RDX: ffff880254e0b340 RSI: ffff8802559b99f8 RDI: ffffffff82181030 [ +0.001413] RBP: ffff880251923b50 R08: 0000000000019d00 R09: ffffffff81385837 [ +0.001431] R10: ffffea0009461200 R11: ffff880253b3fe72 R12: 0000000000000000 [ +0.001411] R13: 0000000000000000 R14: 0000000000000001 R15: ffff88025338c212 [ +0.001436] FS: 00007f7a8db567c0(0000) GS:ffff88025fb00000(0000) knlGS:0000000000000000 [ +0.001431] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ +0.001445] CR2: 0000000000000014 CR3: 0000000251854000 CR4: 00000000001406e0 [ +0.001451] Stack: [ +0.001433] ffff880253302e88 ffff88024d602054 ffff88009879d000 ffff880253302e88 [ +0.001480] ffff880255f7e000 386430425355d030 ff00393133313a63 ffff8800984f4098 [ +0.001466] ffff8800984f4000 ffffffffc0e95728 ffff88009879d030 ffffffffc0e955a0 [ +0.001497] Call Trace: [ +0.001295] [<ffffffffc0067cfd>] usb_probe_interface+0x1bd/0x300 [usbcore] [ +0.001427] [<ffffffff815987e9>] driver_probe_device+0x249/0x450 [ +0.001400] [<ffffffff81598a74>] __driver_attach+0x84/0x90 [ +0.001398] [<ffffffff815989f0>] ? driver_probe_device+0x450/0x450 [ +0.001417] [<ffffffff815963e4>] bus_for_each_dev+0x64/0xa0 [ +0.001369] [<ffffffff81597ede>] driver_attach+0x1e/0x20 [ +0.001396] [<ffffffff815979ab>] bus_add_driver+0x1eb/0x280 [ +0.001365] [<ffffffff81599390>] driver_register+0x60/0xe0 [ +0.001369] [<ffffffffc0066834>] usb_register_driver+0x84/0x140 [usbcore] [ +0.001422] [<ffffffffc0e9d000>] ? 0xffffffffc0e9d000 [ +0.001370] [<ffffffffc0e9d01e>] usb_audio_driver_init+0x1e/0x20 [snd_usb_audio] [ +0.001472] [<ffffffff8100211b>] do_one_initcall+0xab/0x1d0 [ +0.001547] [<ffffffff8115e732>] do_init_module+0x5f/0x1cd [ +0.001469] [<ffffffff810f9ec8>] load_module+0x1368/0x1510 [ +0.001542] [<ffffffff810f75d0>] ? symbol_put_addr+0x50/0x50 [ +0.001468] [<ffffffff810fa2a8>] SyS_finit_module+0xc8/0xf0 [ +0.001451] [<ffffffff81003c6e>] do_syscall_64+0x5e/0xc0 [ +0.001538] [<ffffffff817fb425>] entry_SYSCALL64_slow_path+0x25/0x25 [ +0.001397] Code: f8 01 00 44 89 e6 48 89 df e8 dc 73 00 00 85 c0 89 c1 0f 88 99 00 00 00 48 8b 7b 10 e8 c9 0b 80 c0 85 c0 89 c1 0f 88 86 00 00 00 <41> 80 7d 14 00 0f 85 90 04 00 00 48 63 03 48 89 1c c5 40 90 e9 [ +0.001513] RIP [<ffffffffc0e79921>] usb_audio_probe+0x1e1/0x8f0 [snd_usb_audio] [ +0.001468] RSP <ffff880251923af0> [ +0.001576] CR2: 0000000000000014 [ +0.010610] ---[ end trace c9cdd61401b8f1d9 ]---

3 4

Re: [alsa-devel] snd_hda_core oops with Linux v4.5-6229-gf7813ad5cbfd
by Takashi Iwai 18 Mar '16

18 Mar '16

On Fri, 18 Mar 2016 21:59:51 +0100, Josh Boyer wrote: > > Hi, > > I have a celeron based NUC that is giving me the following backtraces > after Linus merged the sound tree today. This is with the top of the > tree at f7813ad5cbfd as mentioned in the subject. > > The machine has an HDMI port, but it is not connected to any thing. It should have been fixed now on my sound.git tree. Try to pull for-linus branch. I'll send another pull request in early next week. thanks, Takashi

1 0

[alsa-devel] [RFC PATCH 0/3] ASoC: pcm1792a: Add support for pcm1795 and pcm1796
by Michael Trimarchi 18 Mar '16

18 Mar '16

Those patches add support for pcm1795 and pcm1796. Both are compatible with the pcm1792a with a minor change for the pcm1795 Michael Trimarchi (3): ASoC: pcm1792a: Rename internal data and function to pcm179x ASoC: pcm1792a: Rename pcm1792a to pcm179x ASoC: pcm179x: Add support for pcm1795 and pcm1796 .../devicetree/bindings/sound/pcm1792a.txt | 18 -- .../devicetree/bindings/sound/pcm179x.txt | 18 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/pcm1792a.c | 271 ------------------ sound/soc/codecs/pcm1792a.h | 27 -- sound/soc/codecs/pcm179x.c | 314 +++++++++++++++++++++ sound/soc/codecs/pcm179x.h | 31 ++ 8 files changed, 368 insertions(+), 321 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm179x.txt delete mode 100644 sound/soc/codecs/pcm1792a.c delete mode 100644 sound/soc/codecs/pcm1792a.h create mode 100644 sound/soc/codecs/pcm179x.c create mode 100644 sound/soc/codecs/pcm179x.h -- 2.7.0

4 21

[alsa-devel] Playing mono files on a stereo only device
by Champ, Andy 18 Mar '16

18 Mar '16

Hi All, I'm new to this area, so please forgive any too-obvious questions. I'm working on a device driver for an SOC. The audio device is connected via an I2S bus, which is inherently stereo. The driver code that I've inherited (from a similar, but not the same, chip) claimed to support mono - but didn't. I've changed the configuration in the device's snd_soc_dai_driver structure to stereo only (channels_min=channels_max=2) and everything works fine when playing a stereo file. AIUI an Alsa plug should be able to take a mono file, convert it on the fly to stereo, and play that out. I've spent a couple of days playing with it, and not really got anywhere. Does anyone have an example where this is being done on an SOC? Currently the error I am having indicates that it's a problem with access. An IOCTL is being sent by the ALSA code to the kernel, of type SNDRV_PCM_IOCTL_HW_REFINE, requesting an access. This fails. I've not even been able to decide if an access is meant to be a mask or a value. The type is typedef int __bitwise snd_pcm_access_t; which would tend to say it's mask (bitwise?) though I am guessing here, but the values are 0 1 2 3 4 - which says it isn't. Just to really confuse me the mask is logged as 7. Is there anything which explains what this snd_pcm_access_t is? Thanks

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix kbuild dependency
by Vinod Koul 18 Mar '16

18 Mar '16

The Skylake driver selects SND_HDA_I915 which causes kbuild to spew warning: warning: (SND_SOC_INTEL_SKYLAKE) selects SND_HDA_I915 which has unmet direct dependencies (SOUND && !M68K && !UML && SND && DRM_I915 && SND_HDA_CORE) The SND_HDA_I915 should not be selected so drop that. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3e6c2300457..1120f4f4d011 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -163,7 +163,6 @@ config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE select SND_SOC_TOPOLOGY - select SND_HDA_I915 select SND_SOC_INTEL_SST config SND_SOC_INTEL_SKL_RT286_MACH -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix forgotten HDMI monitor_present update
by Takashi Iwai 18 Mar '16

18 Mar '16

We forgot to copy monitor_present value when updating the ELD information. This won't change the ELD retrieval and the jack notification behavior, but appears only in the proc output. In that sense, it's no fatal error, but a bug is a bug is a bug. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8cdb804aa9cf..e7d9453ecd10 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1358,6 +1358,7 @@ static void update_eld(struct hda_codec *codec, eld->eld_size) != 0) eld_changed = true; + pin_eld->monitor_present = eld->monitor_present; pin_eld->eld_valid = eld->eld_valid; pin_eld->eld_size = eld->eld_size; if (eld->eld_valid) -- 2.7.4

1 0

[alsa-devel] [PATCH 5/5] dt/bindings: sound: new chip added into TAS571x binding
by Petr Kulhavy 18 Mar '16

18 Mar '16

This adds the TAS5721 into the TAS571x binding. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- Documentation/devicetree/bindings/sound/tas571x.txt | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt index 0ac31d8..f1aade3 100644 --- a/Documentation/devicetree/bindings/sound/tas571x.txt +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -1,4 +1,4 @@ -Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers +Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers The codec is controlled through an I2C interface. It also has two other signals that can be wired up to GPIOs: reset (strongly recommended), and @@ -6,7 +6,7 @@ powerdown (optional). Required properties: -- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719" +- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719", "ti,tas5721" - reg: The I2C address of the device - #sound-dai-cells: must be equal to 0 @@ -25,6 +25,8 @@ Optional properties: - PVDD_B-supply: regulator phandle for the PVDD_B supply (5711) - PVDD_C-supply: regulator phandle for the PVDD_C supply (5711) - PVDD_D-supply: regulator phandle for the PVDD_D supply (5711) +- DRVDD-supply: regulator phandle for the DRVDD supply (5721) +- PVDD-supply: regulator phandle for the PVDD supply (5721) Example: -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] sst: select DW_DMAC_CORE since it's mandatory
by Andy Shevchenko 18 Mar '16

18 Mar '16

While here group selectors under SND_SOC_INTEL_HASWELL and SND_SOC_INTEL_BAYTRAIL. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/Kconfig | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 05fde5e6e..bc5281e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -35,15 +35,17 @@ config SND_SOC_INTEL_SST_ACPI config SND_SOC_INTEL_HASWELL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST config SND_SOC_INTEL_BAYTRAIL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -55,8 +57,6 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help @@ -66,8 +66,6 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help @@ -76,10 +74,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ - I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST + depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help -- 2.5.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Make sure we have a card when displaying component widgets
by Mark Brown 18 Mar '16

18 Mar '16

The dummy component is reused for all cards so we special case and don't bind it to any of them. This means that code like that displaying the component widgets that tries to look at the card will crash. In the future we will fix this by ensuring that the dummy component looks like other components but that is invasive and so not suitable for a fix. Instead add a special case check here. Reported-by: Harry Pan <harry.pan(a)intel.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a81dfd..c4464858bf01 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2188,6 +2188,13 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, int count = 0; char *state = "not set"; + /* card won't be set for the dummy component, as a spot fix + * we're checking for that case specifically here but in future + * we will ensure that the dummy component looks like others. + */ + if (!cmpnt->card) + return 0; + list_for_each_entry(w, &cmpnt->card->widgets, list) { if (w->dapm != dapm) continue; -- 2.7.0

1 0

Re: [alsa-devel] [Intel-gfx] ✗ Fi.CI.BAT: failure for drm/i915/bxt: Fix off-by-one error in Broxton PLL IDs (rev3)
by Daniel Vetter 18 Mar '16

18 Mar '16

On Wed, Mar 16, 2016 at 02:37:24PM +0200, Tomi Sarvela wrote: > On Wednesday 16 March 2016 10:48:43 Imre Deak wrote: > > Tomi, noticed two things that maybe infrastructure related, see below: > > > > > Test drv_module_reload_basic: > > > skip -> PASS (bdw-nuci7) > > > Test gem_ringfill: > > > Subgroup basic-default-s3: > > > pass -> DMESG-WARN (skl-nuci5) > > > Test gem_storedw_loop: > > > Subgroup basic-bsd: > > > pass -> DMESG-WARN (skl-nuci5) > > > Subgroup basic-bsd1: > > > pass -> DMESG-WARN (skl-nuci5) > > > > Unrelated platform. All of the above are SND resume failures. Ville > > said he contacted Takashi about them. > > SND is recurring problem. I'd like to see drm-intel-nightly not breaking when > pulling sound updates. We need to escalate this again, snd team seriously sucks at quality control :( Adding Libin&Takashi. If this doesn't improve I need to throw sound trees out of -nightly for real. -Daniel -- Daniel Vetter Software Engineer, Intel Corporation http://blog.ffwll.ch

4 9

[alsa-devel] Bad rever-select of CONFIG_SND_HDA_I915
by Takashi Iwai 18 Mar '16

18 Mar '16

Hi, while build testing, I noticed the following warning: warning: (SND_SOC_INTEL_SKYLAKE) selects SND_HDA_I915 which has unmet direct dependencies (SOUND && !M68K && !UML && SND && DRM_I915 && SND_HDA_CORE) And it's indeed broken: you can't select CONFIG_SND_HDA_I915 from the driver side. This is the condition depending on DRM i915 driver. Fixing this doesn't look too trivial, though. Since SND_SOC_INTEL_SKYLAKE is the item to be selected, it can't have "depends on". So, the top-level configs that select SND_SOC_INTEL_SKYLAKE need to have "depends on DRM_I915" lines, or modify the driver to be built without i915 binding. Takashi

2 7

[alsa-devel] [PATCH] ASoC: rt5640: Correct the digital interface data select
by Sugar Zhang 18 Mar '16

18 Mar '16

this patch corrects the interface adc/dac control register definition according to datasheet. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5640.h | 36 ++++++++++++++++++------------------ 2 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 11d032c..48dbb2f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -359,7 +359,7 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, /* Interface data select */ static const char * const rt5640_data_select[] = { - "Normal", "left copy to right", "right copy to left", "Swap"}; + "Normal", "Swap", "left copy to right", "right copy to left"}; static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 83a7150..f84231e 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -442,39 +442,39 @@ #define RT5640_IF1_DAC_SEL_MASK (0x3 << 14) #define RT5640_IF1_DAC_SEL_SFT 14 #define RT5640_IF1_DAC_SEL_NOR (0x0 << 14) -#define RT5640_IF1_DAC_SEL_L2R (0x1 << 14) -#define RT5640_IF1_DAC_SEL_R2L (0x2 << 14) -#define RT5640_IF1_DAC_SEL_SWAP (0x3 << 14) +#define RT5640_IF1_DAC_SEL_SWAP (0x1 << 14) +#define RT5640_IF1_DAC_SEL_L2R (0x2 << 14) +#define RT5640_IF1_DAC_SEL_R2L (0x3 << 14) #define RT5640_IF1_ADC_SEL_MASK (0x3 << 12) #define RT5640_IF1_ADC_SEL_SFT 12 #define RT5640_IF1_ADC_SEL_NOR (0x0 << 12) -#define RT5640_IF1_ADC_SEL_L2R (0x1 << 12) -#define RT5640_IF1_ADC_SEL_R2L (0x2 << 12) -#define RT5640_IF1_ADC_SEL_SWAP (0x3 << 12) +#define RT5640_IF1_ADC_SEL_SWAP (0x1 << 12) +#define RT5640_IF1_ADC_SEL_L2R (0x2 << 12) +#define RT5640_IF1_ADC_SEL_R2L (0x3 << 12) #define RT5640_IF2_DAC_SEL_MASK (0x3 << 10) #define RT5640_IF2_DAC_SEL_SFT 10 #define RT5640_IF2_DAC_SEL_NOR (0x0 << 10) -#define RT5640_IF2_DAC_SEL_L2R (0x1 << 10) -#define RT5640_IF2_DAC_SEL_R2L (0x2 << 10) -#define RT5640_IF2_DAC_SEL_SWAP (0x3 << 10) +#define RT5640_IF2_DAC_SEL_SWAP (0x1 << 10) +#define RT5640_IF2_DAC_SEL_L2R (0x2 << 10) +#define RT5640_IF2_DAC_SEL_R2L (0x3 << 10) #define RT5640_IF2_ADC_SEL_MASK (0x3 << 8) #define RT5640_IF2_ADC_SEL_SFT 8 #define RT5640_IF2_ADC_SEL_NOR (0x0 << 8) -#define RT5640_IF2_ADC_SEL_L2R (0x1 << 8) -#define RT5640_IF2_ADC_SEL_R2L (0x2 << 8) -#define RT5640_IF2_ADC_SEL_SWAP (0x3 << 8) +#define RT5640_IF2_ADC_SEL_SWAP (0x1 << 8) +#define RT5640_IF2_ADC_SEL_L2R (0x2 << 8) +#define RT5640_IF2_ADC_SEL_R2L (0x3 << 8) #define RT5640_IF3_DAC_SEL_MASK (0x3 << 6) #define RT5640_IF3_DAC_SEL_SFT 6 #define RT5640_IF3_DAC_SEL_NOR (0x0 << 6) -#define RT5640_IF3_DAC_SEL_L2R (0x1 << 6) -#define RT5640_IF3_DAC_SEL_R2L (0x2 << 6) -#define RT5640_IF3_DAC_SEL_SWAP (0x3 << 6) +#define RT5640_IF3_DAC_SEL_SWAP (0x1 << 6) +#define RT5640_IF3_DAC_SEL_L2R (0x2 << 6) +#define RT5640_IF3_DAC_SEL_R2L (0x3 << 6) #define RT5640_IF3_ADC_SEL_MASK (0x3 << 4) #define RT5640_IF3_ADC_SEL_SFT 4 #define RT5640_IF3_ADC_SEL_NOR (0x0 << 4) -#define RT5640_IF3_ADC_SEL_L2R (0x1 << 4) -#define RT5640_IF3_ADC_SEL_R2L (0x2 << 4) -#define RT5640_IF3_ADC_SEL_SWAP (0x3 << 4) +#define RT5640_IF3_ADC_SEL_SWAP (0x1 << 4) +#define RT5640_IF3_ADC_SEL_L2R (0x2 << 4) +#define RT5640_IF3_ADC_SEL_R2L (0x3 << 4) /* REC Left Mixer Control 1 (0x3b) */ #define RT5640_G_HP_L_RM_L_MASK (0x7 << 13) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Do not traverse widget hooks to snd-soc-dummy
by Harry Pan 18 Mar '16

18 Mar '16

All components are initially given an empty card when registering platform, and since the commit 6e78108bda78 ("ASoC: core: Don't probe the component which is dummy")', snd-soc-dummy will not be probed so that it remains an empty card assigned. This patch ignores to iterate widget hooks to the 'snd-soc-dummy' component, else it will trigger memory fault because of invalid dereference address of card->widgets. Test by grep -rsI "hello" /sys/devices/platform/skl_nau88l25_ssm4567_i2s/ Conflicts: sound/soc/soc-dapm.c Signed-off-by: Harry Pan <harry.pan(a)intel.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 581175a..0bc15c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2188,6 +2188,9 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, int count = 0; char *state = "not set"; + if (!strcmp(cmpnt->name, "snd-soc-dummy")) + return 0; + list_for_each_entry(w, &cmpnt->card->widgets, list) { if (w->dapm != dapm) continue; -- 2.1.2

5 7

Re: [alsa-devel] [pulseaudio-discuss] ASoC and pulseaudio
by Liam Girdwood 17 Mar '16

17 Mar '16

+ alsa, Takashi and Mark since it's a naming convention being discussed as part of this work. On Mon, 2016-03-14 at 08:17 +0000, Lu, Han wrote: > Hi, > > > -----Original Message----- > > From: pulseaudio-discuss [mailto:pulseaudio-discuss- > > bounces(a)lists.freedesktop.org] On Behalf Of Liam Girdwood > > Sent: Tuesday, February 23, 2016 7:13 PM > > To: General PulseAudio Discussion <pulseaudio- > > discuss(a)lists.freedesktop.org> > > Cc: Linux Upstreaming Team <linux(a)endlessm.com>; han.lu > > <han.lu(a)linux.intel.com> > > Subject: Re: [pulseaudio-discuss] ASoC and pulseaudio > > > > On Sat, 2016-02-20 at 12:22 -0600, Pierre-Louis Bossart wrote: > > > r > > > On 02/17/2016 09:27 AM, Carlo Caione wrote: > > > > Hi, > > > > In our daily work we are seeing more and more laptops coming in with > > > > SoC audio (ASoC). In the most recent case, we are working with a new > > > > consumer laptop based on Intel Cherry Trail. As you probably know to > > > > have audio working on these laptops / codecs a fairly lengthy hw > > > > specific mixer configuration is needed. > > > > The problem we are facing is how to ship a single PA / ALSA package > > > > containing a working configuration for several of those laptops and > > > > we were wondering what is the most correct / upstream-friendly way > > > > to do that. > > > > > > > > I'm already excluding here some trivial workarounds like: shipping a > > > > different PA package for each hw or using an hw-tailored script to > > > > load the correct ALSA configuration. > > > > > > > > The first solution we came up with is just to write a profile-set > > > > configuration file loaded by udev for each cards / laptop with the > > > > correct mixer configuration already embedded in the path > > > > configuration files. This looks a bit like an overkill when you have > > > > to change tens of controls most of which just one time (once you > > > > have setup the codec selecting the main path is usually just matter > > > > of changing a few controls to change the output / input path). > > > > > > > > Another possibility is to use UCM and let PA create automatically > > > > from that profile and path configuration files. This looks promising > > > > but we were a bit worried about the UCM support in PA and how > > > > stable/adopted is UCM itself. > > > > Fwiw, I've not had any problems using PA and UCM together on my BDW + > > RT286 based development platforms. The problem we have atm is matching > > and quirks as Pierre describes. > > > > > > > > > > So, what is the suggested way to accomplish this and in general how > > > > PA is trying to address this problem? I expect in the future to see > > > > many more ASoC coming into the laptops world, how will the community > > > > make it so that you install a distro and then sound "just works"? > > > > > > Not an easy problem. I generated a set of UCM files for baytrail-based > > platforms and I was planning on doing the same for CHT now that there are a > > set of commercially available devices. > > > There are however a set of limitations that are known already, e.g. > > > the UCM file matches the machine driver name, so if there are any quirks > > inside the driver we'd need to add a DMI-based info to the card name and > > also build UCM files based on an include mechanism, otherwise it's going to > > be copy/paste hell Adding Liam to make sure he sees this thread. > > > > > > > Fwiw, we have been looking at providing some fixes for naming and quirks. > > We should probably schedule the work now. > > > > For naming we have the problem of duplication in the driver name e.g. > > > > cat /proc/asound/cards > > 0 [bytrt5640 ]: byt-rt5640 - byt-rt5640 > > byt-rt5640 > > > > The driver name actually has three strings iirc and we could differentiate here > > (by adding DMI name, plus any others?? ) to help UCM and PA deal with any > > quirks. > > > > UCM also does not currently support #include. The intention is to provide a > > method to define mixer settings on a codec per codec basis and the these > > could be #included into a machine UCM file. The machine UCM file would > > also define settings for any quirks and would #include any codec UCM files > > that were needed. PA would load the machine UCM file (based on > > improvements to the driver naming above) intead of the generic UCM file > > that is loaded atm. > > > > Liam > > > > We can fix this from driver level and user space level: > (1) to reporting more information from driver. > ALSA sound cards have 3 names. longname, shortname and drivername. e.g. > from soc-core.c :- > > snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), > "%s", card->name); > snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), > "%s", card->long_name ? card->long_name : card->name); > snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), > "%s", card->driver_name ? card->driver_name : card->name); > > We can additionally set the long_name and driver_name in ASoC if they were not set. > > The naming scheme could be :- > > a) shortname is current machine driver name i.e. "byt-rt5640" > b) longname is DMI boardname + 1 i.e. "Minnowboard Max: byt-rt5640" > c) driver name is platform driver name i.e. "baytrail-pcm-audio" > > (2) to enable UCM to support "include" parsing, so each codec will have its own UCM file to define all the use cases for the codec only, and the codec UCM file can be included and the codec use cases can be referenced by different machine driver UCM files. > > I'll try to work out RFC patches to implement the features above. > > BR, > Han > > > > > > > > _______________________________________________ > > > pulseaudio-discuss mailing list > > > pulseaudio-discuss(a)lists.freedesktop.org > > > https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss > > > > > > _______________________________________________ > > pulseaudio-discuss mailing list > > pulseaudio-discuss(a)lists.freedesktop.org > > https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss > _______________________________________________ > pulseaudio-discuss mailing list > pulseaudio-discuss(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss

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[alsa-devel] [PATCH] ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
by Takashi Sakamoto 17 Mar '16

17 Mar '16

In control compatibility layer, when no elements are found by ELEM_READ/ELEM_WRITE ioctl commands, ENXIO is returned. On the other hand, in core implementation, ENOENT is returned. This is not good for ALSA ctl applications. This commit changes the return value from the compatibility layer so that the same value is returned. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control_compat.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 0608f21..1fa7076 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -196,7 +196,7 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, kctl = snd_ctl_find_id(card, id); if (! kctl) { up_read(&card->controls_rwsem); - return -ENXIO; + return -ENOENT; } info = kzalloc(sizeof(*info), GFP_KERNEL); if (info == NULL) { -- 2.7.0

2 1

[alsa-devel] adau1701: Different sample rates need different firmwares
by Pascal Huerst 17 Mar '16

17 Mar '16

Hey Lars, I'm having an issue with the adau1701 codec. There is this line which checks, if the mclk/lrclk ratio has changed and whether it is necessary to reset the codec and upload a new firmware. http://lxr.free-electrons.com/source/sound/soc/codecs/adau1701.c#L444 But if we change mclk in the machine driver (for example to playback 44.1), this ration might not change, but still we have to upload a new firmware, so shouldn't we probably be checking for sample rate changes instead: if (adau1701->current_rate != params_rate(params) { adau1701->current_rate = params_rate(params); ret = adau1701_reset(codec, clkdiv, params_rate(params)); if (ret < 0) return ret; } Or did I miss something? regards pascal

2 1

Re: [alsa-devel] different return value from some ioctl commands for x86 binary executed on x86_64
by Takashi Iwai 17 Mar '16

17 Mar '16

On Thu, 17 Mar 2016 04:52:52 +0100, Takashi Sakamoto wrote: > > Hi, > > When testing ioctl compatibility layer for ALSA ctl interface, I found > that error code of some ioctl commands are different depending on binary > architecture. > > The commands are SNDRV_CTL_IOCTL_ELEM_READ/SNDRV_CTL_IOCTL_ELEM_WRITE. > When passing data for non-existent element, ENOENT is returned by the > core implementation. On the other hand, ENXIO is returned by the > compatibility layer. > > This code is for reproduction of this bug. > > #include <stdio.h> > #include <stdlib.h> > #include <stdbool.h> > #include <sys/types.h> > #include <sys/stat.h> > #include <fcntl.h> > #include <sys/ioctl.h> > #include <errno.h> > #include <string.h> > > #include <sound/asound.h> > > static void check_elem_read(int fd) > { > struct snd_ctl_elem_value data = {0}; > if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_READ, &data) < 0) { > printf("ioctl(SNDRV_CTL_IOCTL_ELEM_READ): %s\n", > strerror(errno)); > } > } > > static void check_elem_write(int fd) > { > struct snd_ctl_elem_value data = {0}; > if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &data) < 0) { > printf("ioctl(SNDRV_CTL_IOCTL_ELEM_WRITE): %s\n", > strerror(errno)); > } > } > > int main(void) > { > int fd; > > fd = open("/dev/snd/controlC0", O_RDONLY); > if (fd < 0) { > printf("open(2): %s\n", strerror(errno)); > return EXIT_FAILURE; > } > > check_elem_read(fd); > check_elem_write(fd); > > return EXIT_SUCCESS; > } > > I tested on x86_64 machine with enough support for multi-arch. > > $ gcc -Wall -m64 -o ./test ./test.c ; ./test > ioctl(SNDRV_CTL_IOCTL_ELEM_READ): No such file or directory > ioctl(SNDRV_CTL_IOCTL_ELEM_WRITE): No such file or directory > > $ gcc -Wall -m32 -o ./test ./test.c ; ./test > ioctl(SNDRV_CTL_IOCTL_ELEM_READ): No such device or address > ioctl(SNDRV_CTL_IOCTL_ELEM_WRITE): No such device or address > > > This bug is due to return value of condition statement for > 'snd_ctl_find()'. I wrote a patch to fix it. > > > >From d7a8fd14a1bc100dbaf9bf28af47094d5ccb882b Mon Sep 17 00:00:00 2001 > From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > Date: Wed, 16 Mar 2016 13:40:31 +0900 > Subject: [PATCH] ALSA: ctl: change error code in compatibility layer so that > it's the same code as core implementation > > In control compatibility layer, when no elements are found by > ELEM_READ/ELEM_WRITE ioctl commands, ENXIO is returned. On the other > hand, in core implementation, ENOENT is returned. This is not good for > applications. > > This commit changes the return value from the compatibility layer so > that the same error code is returned. > > diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c > index 0608f21..1fa7076 100644 > --- a/sound/core/control_compat.c > +++ b/sound/core/control_compat.c > @@ -196,7 +196,7 @@ static int get_ctl_type(struct snd_card *card, > struct snd_ctl_elem_id *id, > kctl = snd_ctl_find_id(card, id); > if (! kctl) { > up_read(&card->controls_rwsem); > - return -ENXIO; > + return -ENOENT; > } > info = kzalloc(sizeof(*info), GFP_KERNEL); > if (info == NULL) { > > Well, it's already in merge window for Linux 4.6. I don't mind to > postpone this patch to next developing cycle, while I also think it > better to fix the bug now. So I leave the decision to maintainers. Looks good to me, and trivial enough to be merged for 4.6. Please resubmit with a proper sign off. thanks, Takashi

2 1

[alsa-devel] [PATCH v7 0/6] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 17 Mar '16

17 Mar '16

Not much have changed since v6[1] patch series. I did everything addressed every comment, mostly by robots :), to previous series and there was one bug fixed in tda998x side. Unless there is any significant reaction to this series, I am going to put it to rest for a while now. I have other things to do. There is currently two other patch series[2][3] that depend on the first two (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 two 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 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 (6): ALSA: pcm: add IEC958 channel status helper for hw_params 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 | 276 ++++++++++++-- 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 | 411 +++++++++++++++++++++ 12 files changed, 915 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

2 7

[alsa-devel] [PATCH v3 00/22] Sharing media resources across ALSA and au0828 drivers
by Shuah Khan 17 Mar '16

17 Mar '16

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API and Media Controller API to share media resource (tuner). This Patch v3 series is based on linux_media master. This work addresses Mauro and Takashi's comments on Patch v2 series. Changes since Patch v2 series: - Added documentation for uapi and new MC interfaces 1. Ran make for documentation and Mauro's doc script. Thanks for sharing the script. - Updated commit logs as needed for clarity. - media: v4l-core add enable/disable source common interfaces Addressed comments - added static inlines in header file - media: dvb-frontend invoke enable/disable_source handlers Moved the patch after enable/disable handlers and left pipe in dvb_frontend_private - media: au0828 video remove au0828_enable_analog_tuner() and media: au0828 video change to use v4l_enable_media_source() 1. Collapsed these two patches - it doesn't make sense to keep them separate. Also added FIXME as per Mauro's suggestion. - media: au0828 change to use Managed Media Controller API 1. Added comment explaining the need for using udev->product instead of dev->board.name - media: au0828 add enable, disable source handlers 1. Changed to handle S-Video and Composite inputs. 2. Fixed a bug that lets Video in when ALSA holds the source. 3. Added comments to one FIXME for s_input changes. Fix is in progress and will send it in a day or two combined with other issues found in au0828 video in the way it handles input. Doesn't impact this routine, changes are all in au0828-video. - sound/usb: Use Media Controller API to share media resources 1. Addressed comments from Mauro and Takashi about void pointer use. 2. Fixed kbuildbot error when MEDIA_SUPPORT=m - Dropped media: Ensure media device unregister is done only once Device removal worked without any changes to media unregister path. This work isn't necessary. Shuah Khan (22): [media] Docbook: media-types.xml: Add ALSA Media Controller Intf types uapi/media.h: Declare interface types for ALSA [media] Docbook: media-types.xml: Add Audio Function Entities media: Add ALSA Media Controller function entities media: Media Controller register/unregister entity_notify API media: Media Controller enable/disable source handler API media: Media Controller export non locking __media_entity_setup_link() edia: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add enable/disable source common interfaces media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: Change v4l-core to check if source is free media: au0828 change to use Managed Media Controller API media: au0828 handle media_init and media_register window media: au0828 create tuner to decoder link in disabled state media: au0828 disable tuner to demod link media: au0828-core register entity_notify hook media: au0828 add enable, disable source handlers media: dvb-frontend invoke enable/disable_source handlers media: au0828 video change to use v4l_enable_media_source() sound/usb: Use Media Controller API to share media resources Documentation/DocBook/media/v4l/media-types.xml | 52 ++++ drivers/media/dvb-core/dvb_frontend.c | 135 ++-------- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 42 +++ drivers/media/media-entity.c | 51 +++- drivers/media/usb/au0828/au0828-core.c | 325 ++++++++++++++++++++++-- drivers/media/usb/au0828/au0828-video.c | 102 +++----- drivers/media/usb/au0828/au0828.h | 6 + drivers/media/v4l2-core/Makefile | 1 + drivers/media/v4l2-core/v4l2-fh.c | 2 + drivers/media/v4l2-core/v4l2-ioctl.c | 30 +++ drivers/media/v4l2-core/v4l2-mc.c | 52 ++++ drivers/media/v4l2-core/videobuf2-core.c | 4 + include/media/media-device.h | 86 +++++++ include/media/media-entity.h | 19 ++ include/media/v4l2-dev.h | 1 + include/media/v4l2-mc.h | 65 +++++ include/uapi/linux/media.h | 17 ++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 + sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++ sound/usb/media.h | 72 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 +- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 31 files changed, 1240 insertions(+), 220 deletions(-) create mode 100644 drivers/media/v4l2-core/v4l2-mc.c create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.5.0

5 46

[alsa-devel] [PATCH] sound/usb: Fix memory leak in media_snd_stream_delete() during unbind
by Shuah Khan 17 Mar '16

17 Mar '16

media_snd_stream_delete() fails to release resources during unbind. This leads to use-after-free in media_gobj_create() on a subsequent bind. [ 1445.086410] BUG: KASAN: use-after-free in media_gobj_create+0x3a1/0x470 [media] at addr ffff8801ead49998 [ 1445.086771] Call Trace: [ 1445.086779] [<ffffffff81ade373>] dump_stack+0x67/0x94 [ 1445.086785] [<ffffffff81523c29>] print_trailer+0xf9/0x150 [ 1445.086790] [<ffffffff81529644>] object_err+0x34/0x40 [ 1445.086796] [<ffffffff8152b9e1>] kasan_report_error+0x221/0x530 [ 1445.086803] [<ffffffff8152bfb3>] __asan_report_store8_noabort+0x43/0x50 [ 1445.086813] [<ffffffffa0a79341>] ? media_gobj_create+0x3a1/0x470 [media] [ 1445.086822] [<ffffffffa0a79341>] media_gobj_create+0x3a1/0x470 [media] [ 1445.086831] [<ffffffffa0a705a9>] media_device_register_entity+0x259/0x6f0 [media] [ 1445.086841] [<ffffffffa0a70350>] ? media_device_unregister_entity_notify+0x100/0x100 [media] [ 1445.086846] [<ffffffff81526232>] ? ___slab_alloc+0x172/0x500 [ 1445.086854] [<ffffffff81203548>] ? mark_held_locks+0xc8/0x120 [ 1445.086859] [<ffffffff81526610>] ? __slab_alloc+0x50/0x70 [ 1445.086878] [<ffffffffa0fe711c>] ? media_snd_mixer_init+0x16c/0x500 [snd_usb_audio] [ 1445.086884] [<ffffffff8152b086>] ? kasan_unpoison_shadow+0x36/0x50 [ 1445.086890] [<ffffffff8152b086>] ? kasan_unpoison_shadow+0x36/0x50 [ 1445.086895] [<ffffffff8152b0fe>] ? kasan_kmalloc+0x5e/0x70 Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- sound/usb/media.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index 44a5de9..0d03773 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -135,7 +135,7 @@ void media_snd_stream_delete(struct snd_usb_substream *subs) if (mctl && mctl->media_dev) { struct media_device *mdev; - mdev = subs->stream->chip->media_dev; + mdev = mctl->media_dev; if (mdev && media_devnode_is_registered(&mdev->devnode)) { media_devnode_remove(mctl->intf_devnode); media_device_unregister_entity(&mctl->media_entity); -- 2.5.0

2 2

[alsa-devel] [PATCH] ASoC: nau8825: support different clock source for FLL function
by John Hsu 17 Mar '16

17 Mar '16

Extend FLL clock source selection. The source can be from MCLK, BCLK or FS. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 82 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 8 +++++ 2 files changed, 67 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 1c87299..ac264ec 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1053,6 +1053,37 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + nau8825->mclk = devm_clk_get(nau8825->dev, "mclk"); + if (IS_ERR(nau8825->mclk)) { + dev_info(nau8825->dev, "No 'mclk' clock found, assume MCLK is managed externally"); + return 0; + } + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + nau8825->mclk_freq = freq; + } + + return 0; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -1064,29 +1095,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1094,7 +1105,32 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8ceb5f3..ed0d8f3 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -113,6 +113,11 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) @@ -320,6 +325,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 2.6.4

3 4

[alsa-devel] [PATCH] ALSA: hda - codec add DP MST support for connection list
by libin.yang＠linux.intel.com 17 Mar '16

17 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patches adds the support of connection list for DP MST. With this, hdmi driver in DP MST mode can easily reuse the connection list mechanism. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 52 +++++++++++++++++++++++++++--------------- sound/pci/hda/hda_codec.h | 15 ++++++------ sound/pci/hda/hda_generic.c | 18 +++++++-------- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/patch_analog.c | 18 ++++++++------- sound/pci/hda/patch_hdmi.c | 8 ++++--- sound/pci/hda/patch_realtek.c | 20 ++++++++-------- sound/pci/hda/patch_sigmatel.c | 4 ++-- sound/pci/hda/patch_via.c | 10 ++++---- 9 files changed, 84 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8374188..d17e38e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -110,23 +110,24 @@ struct hda_conn_list { struct list_head list; int len; hda_nid_t nid; + int dev_id; hda_nid_t conns[0]; }; /* look up the cached results */ static struct hda_conn_list * -lookup_conn_list(struct hda_codec *codec, hda_nid_t nid) +lookup_conn_list(struct hda_codec *codec, hda_nid_t nid, int dev_id) { struct hda_conn_list *p; list_for_each_entry(p, &codec->conn_list, list) { - if (p->nid == nid) + if ((p->nid == nid) && (p->dev_id == dev_id)) return p; } return NULL; } -static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, - const hda_nid_t *list) +static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, + int dev_id, int len, const hda_nid_t *list) { struct hda_conn_list *p; @@ -135,6 +136,7 @@ static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, return -ENOMEM; p->len = len; p->nid = nid; + p->dev_id = dev_id; memcpy(p->conns, list, len * sizeof(hda_nid_t)); list_add(&p->list, &codec->conn_list); return 0; @@ -150,8 +152,13 @@ static void remove_conn_list(struct hda_codec *codec) } } -/* read the connection and add to the cache */ -static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) +/* + * read the connection and add to the cache + * the caller should select the device entry by sending the + * corresponding verb if necessary before calling this function + */ +static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid, + int dev_id) { hda_nid_t list[32]; hda_nid_t *result = list; @@ -166,7 +173,8 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) len = snd_hda_get_raw_connections(codec, nid, result, len); } if (len >= 0) - len = snd_hda_override_conn_list(codec, nid, len, result); + len = snd_hda_override_conn_list(codec, nid, dev_id, + len, result); if (result != list) kfree(result); return len; @@ -176,6 +184,7 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) * snd_hda_get_conn_list - get connection list * @codec: the HDA codec * @nid: NID to parse + * @dev_id: device entry id * @listp: the pointer to store NID list * * Parses the connection list of the given widget and stores the pointer @@ -188,7 +197,7 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) * concurrently, protect with a mutex appropriately. */ int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, - const hda_nid_t **listp) + int dev_id, const hda_nid_t **listp) { bool added = false; @@ -197,7 +206,7 @@ int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, const struct hda_conn_list *p; /* if the connection-list is already cached, read it */ - p = lookup_conn_list(codec, nid); + p = lookup_conn_list(codec, nid, dev_id); if (p) { if (listp) *listp = p->conns; @@ -206,7 +215,7 @@ int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, if (snd_BUG_ON(added)) return -EINVAL; - err = read_and_add_raw_conns(codec, nid); + err = read_and_add_raw_conns(codec, nid, dev_id); if (err < 0) return err; added = true; @@ -218,6 +227,7 @@ EXPORT_SYMBOL_GPL(snd_hda_get_conn_list); * snd_hda_get_connections - copy connection list * @codec: the HDA codec * @nid: NID to parse + * @dev_id: device entry id * @conn_list: connection list array; when NULL, checks only the size * @max_conns: max. number of connections to store * @@ -227,10 +237,11 @@ EXPORT_SYMBOL_GPL(snd_hda_get_conn_list); * Returns the number of connections, or a negative error code. */ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns) + int dev_id, hda_nid_t *conn_list, + int max_conns) { const hda_nid_t *list; - int len = snd_hda_get_conn_list(codec, nid, &list); + int len = snd_hda_get_conn_list(codec, nid, dev_id, &list); if (len > 0 && conn_list) { if (len > max_conns) { @@ -249,6 +260,7 @@ EXPORT_SYMBOL_GPL(snd_hda_get_connections); * snd_hda_override_conn_list - add/modify the connection-list to cache * @codec: the HDA codec * @nid: NID to parse + * @dev_id: device entry id * @len: number of connection list entries * @list: the list of connection entries * @@ -257,18 +269,18 @@ EXPORT_SYMBOL_GPL(snd_hda_get_connections); * * Returns zero or a negative error code. */ -int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, - const hda_nid_t *list) +int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, + int dev_id, int len, const hda_nid_t *list) { struct hda_conn_list *p; - p = lookup_conn_list(codec, nid); + p = lookup_conn_list(codec, nid, dev_id); if (p) { list_del(&p->list); kfree(p); } - return add_conn_list(codec, nid, len, list); + return add_conn_list(codec, nid, dev_id, len, list); } EXPORT_SYMBOL_GPL(snd_hda_override_conn_list); @@ -277,6 +289,7 @@ EXPORT_SYMBOL_GPL(snd_hda_override_conn_list); * @codec: the HDA codec * @mux: NID containing the list * @nid: NID to select + * @dev_id: device entry id * @recursive: 1 when searching NID recursively, otherwise 0 * * Parses the connection list of the widget @mux and checks whether the @@ -284,12 +297,12 @@ EXPORT_SYMBOL_GPL(snd_hda_override_conn_list); * Otherwise it returns -1. */ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, - hda_nid_t nid, int recursive) + hda_nid_t nid, int dev_id, int recursive) { const hda_nid_t *conn; int i, nums; - nums = snd_hda_get_conn_list(codec, mux, &conn); + nums = snd_hda_get_conn_list(codec, mux, dev_id, &conn); for (i = 0; i < nums; i++) if (conn[i] == nid) return i; @@ -304,7 +317,8 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i])); if (type == AC_WID_PIN || type == AC_WID_AUD_OUT) continue; - if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0) + if (snd_hda_get_conn_index(codec, conn[i], nid, dev_id, + recursive) >= 0) return i; } return -1; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 373fcad..4852c1a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -329,11 +329,12 @@ snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, #define snd_hda_get_sub_nodes(codec, nid, start_nid) \ snd_hdac_get_sub_nodes(&(codec)->core, nid, start_nid) int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns); + int dev_id, hda_nid_t *conn_list, + int max_conns); static inline int -snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) +snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid, int dev_id) { - return snd_hda_get_connections(codec, nid, NULL, 0); + return snd_hda_get_connections(codec, nid, dev_id, NULL, 0); } #define snd_hda_get_raw_connections(codec, nid, list, max_conns) \ @@ -341,12 +342,12 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) #define snd_hda_get_num_raw_conns(codec, nid) \ snd_hdac_get_connections(&(codec)->core, nid, NULL, 0); -int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, +int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, int dev_id, const hda_nid_t **listp); -int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, - const hda_nid_t *list); +int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, + int dev_id, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, - hda_nid_t nid, int recursive); + hda_nid_t nid, int dev_id, int recursive); int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, u8 *dev_list, int max_devices); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ca5b89..f8300b4 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -338,7 +338,7 @@ static bool is_reachable_path(struct hda_codec *codec, { if (!from_nid || !to_nid) return false; - return snd_hda_get_conn_index(codec, to_nid, from_nid, true) >= 0; + return snd_hda_get_conn_index(codec, to_nid, from_nid, 0, true) >= 0; } /* nid, dir and idx */ @@ -397,7 +397,7 @@ static bool __parse_nid_path(struct hda_codec *codec, else if (to_nid == (hda_nid_t)(-anchor_nid)) return false; /* hit the exclusive nid */ - nums = snd_hda_get_conn_list(codec, to_nid, &conn); + nums = snd_hda_get_conn_list(codec, to_nid, 0, &conn); for (i = 0; i < nums; i++) { if (conn[i] != from_nid) { /* special case: when from_nid is 0, @@ -696,9 +696,9 @@ static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, int dir) return true; if (dir != HDA_INPUT || get_wcaps_type(wcaps) != AC_WID_AUD_MIX) return false; - if (snd_hda_get_num_conns(codec, nid) != 1) + if (snd_hda_get_num_conns(codec, nid, 0) != 1) return false; - if (snd_hda_get_connections(codec, nid, &conn, 1) < 0) + if (snd_hda_get_connections(codec, nid, 0, &conn, 1) < 0) return false; return !!(get_wcaps(codec, conn) & AC_WCAP_STEREO); } @@ -791,7 +791,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, int type; hda_nid_t nid = path->path[i]; - nums = snd_hda_get_conn_list(codec, nid, &conn); + nums = snd_hda_get_conn_list(codec, nid, 0, &conn); type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_PIN || (type == AC_WID_AUD_IN && codec->single_adc_amp)) { @@ -1058,7 +1058,7 @@ static int add_sw_ctl(struct hda_codec *codec, const char *pfx, int cidx, val = amp_val_replace_channels(val, chs); if (get_amp_direction_(val) == HDA_INPUT) { hda_nid_t nid = get_amp_nid_(val); - int nums = snd_hda_get_num_conns(codec, nid); + int nums = snd_hda_get_num_conns(codec, nid, 0); if (nums > 1) { type = HDA_CTL_BIND_MUTE; val |= nums << 19; @@ -3004,7 +3004,7 @@ static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, const hda_nid_t *list; hda_nid_t nid; - idx = snd_hda_get_conn_index(codec, mix_nid, pin, true); + idx = snd_hda_get_conn_index(codec, mix_nid, pin, 0, true); if (idx < 0) return false; @@ -3017,7 +3017,7 @@ static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, return true; /* check leaf node */ - num_conns = snd_hda_get_conn_list(codec, mix_nid, &list); + num_conns = snd_hda_get_conn_list(codec, mix_nid, 0, &list); if (num_conns < idx) return false; nid = list[idx]; @@ -4760,7 +4760,7 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) const hda_nid_t *conn; bool has_amp; - nums = snd_hda_get_conn_list(codec, mix, &conn); + nums = snd_hda_get_conn_list(codec, mix, 0, &conn); has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 033aa84..c747183 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -636,7 +636,7 @@ static void print_conn_list(struct snd_info_buffer *buffer, } /* Get Cache connections info */ - cache_len = snd_hda_get_conn_list(codec, nid, &list); + cache_len = snd_hda_get_conn_list(codec, nid, 0, &list); if (cache_len >= 0 && (cache_len != conn_len || memcmp(list, conn, conn_len) != 0)) { snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e0fb8c6..6f007ff 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -466,7 +466,7 @@ static int ad1983_auto_smux_enum_info(struct snd_kcontrol *kcontrol, static const char * const texts2[] = { "PCM", "ADC" }; static const char * const texts3[] = { "PCM", "ADC1", "ADC2" }; hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = snd_hda_get_num_conns(codec, dig_out, 0); if (num_conns == 2) return snd_hda_enum_helper_info(kcontrol, uinfo, 2, texts2); @@ -493,7 +493,7 @@ static int ad1983_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = snd_hda_get_num_conns(codec, dig_out, 0); if (val >= num_conns) return -EINVAL; @@ -521,7 +521,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) if (!dig_out) return 0; - num_conns = snd_hda_get_num_conns(codec, dig_out); + num_conns = snd_hda_get_num_conns(codec, dig_out, 0); if (num_conns != 2 && num_conns != 3) return 0; if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &ad1983_auto_smux_mixer)) @@ -546,8 +546,10 @@ static int patch_ad1983(struct hda_codec *codec) set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); /* limit the loopback routes not to confuse the parser */ - snd_hda_override_conn_list(codec, 0x0c, ARRAY_SIZE(conn_0c), conn_0c); - snd_hda_override_conn_list(codec, 0x0d, ARRAY_SIZE(conn_0d), conn_0d); + snd_hda_override_conn_list(codec, 0x0c, 0, ARRAY_SIZE(conn_0c), + conn_0c); + snd_hda_override_conn_list(codec, 0x0d, 0, ARRAY_SIZE(conn_0d), + conn_0d); err = ad198x_parse_auto_config(codec, false); if (err < 0) @@ -745,7 +747,7 @@ static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, static const char * const texts[] = { "PCM", "ADC1", "ADC2", "ADC3", }; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = snd_hda_get_num_conns(codec, 0x0b, 0) + 1; if (num_conns > 4) num_conns = 4; return snd_hda_enum_helper_info(kcontrol, uinfo, num_conns, texts); @@ -768,7 +770,7 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; struct nid_path *path; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = snd_hda_get_num_conns(codec, 0x0b, 0) + 1; if (val >= num_conns) return -EINVAL; @@ -856,7 +858,7 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec) get_wcaps_type(get_wcaps(codec, 0x1d)) != AC_WID_AUD_MIX) return 0; - num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + num_conns = snd_hda_get_num_conns(codec, 0x0b, 0) + 1; if (num_conns != 3 && num_conns != 4) return 0; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index fe4141c..7e03200 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1719,7 +1719,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return -EINVAL; } - per_pin->num_mux_nids = snd_hda_get_connections(codec, pin_nid, + per_pin->num_mux_nids = snd_hda_get_connections(codec, pin_nid, 0, per_pin->mux_nids, HDA_MAX_CONNECTIONS); @@ -2818,14 +2818,16 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, hda_nid_t conns[4]; int nconns; - nconns = snd_hda_get_connections(codec, nid, conns, ARRAY_SIZE(conns)); + nconns = snd_hda_get_connections(codec, nid, 0, conns, + ARRAY_SIZE(conns)); if (nconns == spec->num_cvts && !memcmp(conns, spec->cvt_nids, spec->num_cvts * sizeof(hda_nid_t))) return; /* override pins connection list */ codec_dbg(codec, "hdmi: haswell: override pin connection 0x%x\n", nid); - snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); + snd_hda_override_conn_list(codec, nid, 0, spec->num_cvts, + spec->cvt_nids); } #define INTEL_VENDOR_NID 0x08 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 93d2156..fb70f18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1851,17 +1851,17 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, /* fake the connections during parsing the tree */ hda_nid_t conn1[2] = { 0x0c, 0x0d }; hda_nid_t conn2[2] = { 0x0e, 0x0f }; - snd_hda_override_conn_list(codec, 0x14, 2, conn1); - snd_hda_override_conn_list(codec, 0x15, 2, conn1); - snd_hda_override_conn_list(codec, 0x18, 2, conn2); - snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + snd_hda_override_conn_list(codec, 0x14, 0, 2, conn1); + snd_hda_override_conn_list(codec, 0x15, 0, 2, conn1); + snd_hda_override_conn_list(codec, 0x18, 0, 2, conn2); + snd_hda_override_conn_list(codec, 0x1a, 0, 2, conn2); } else if (action == HDA_FIXUP_ACT_PROBE) { /* restore the connections */ hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; - snd_hda_override_conn_list(codec, 0x14, 5, conn); - snd_hda_override_conn_list(codec, 0x15, 5, conn); - snd_hda_override_conn_list(codec, 0x18, 5, conn); - snd_hda_override_conn_list(codec, 0x1a, 5, conn); + snd_hda_override_conn_list(codec, 0x14, 0, 5, conn); + snd_hda_override_conn_list(codec, 0x15, 0, 5, conn); + snd_hda_override_conn_list(codec, 0x18, 0, 5, conn); + snd_hda_override_conn_list(codec, 0x1a, 0, 5, conn); } } @@ -4607,8 +4607,8 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, make sure 0x14 (front speaker) and 0x15 (headphones) use the stereo DAC, while leaving 0x17 (bass speaker) for node 0x03. */ hda_nid_t conn1[2] = { 0x0c }; - snd_hda_override_conn_list(codec, 0x14, 1, conn1); - snd_hda_override_conn_list(codec, 0x15, 1, conn1); + snd_hda_override_conn_list(codec, 0x14, 0, 1, conn1); + snd_hda_override_conn_list(codec, 0x15, 0, 1, conn1); } } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 37b70f8..5bff793 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -984,7 +984,7 @@ static int stac_create_spdif_mux_ctls(struct hda_codec *codec) if (cfg->dig_outs < 1) return 0; - num_cons = snd_hda_get_num_conns(codec, cfg->dig_out_pins[0]); + num_cons = snd_hda_get_num_conns(codec, cfg->dig_out_pins[0], 0); if (num_cons <= 1) return 0; @@ -4543,7 +4543,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; - num_dacs = snd_hda_get_num_conns(codec, 0x0a) - 1; + num_dacs = snd_hda_get_num_conns(codec, 0x0a, 0) - 1; if (num_dacs < 3 || num_dacs > 5) { codec_warn(codec, "Could not determine number of channels defaulting to DAC count\n"); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fc30d1e..14d9010 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -867,7 +867,7 @@ static int add_secret_dac_path(struct hda_codec *codec) if (!spec->gen.mixer_nid) return 0; - nums = snd_hda_get_connections(codec, spec->gen.mixer_nid, conn, + nums = snd_hda_get_connections(codec, spec->gen.mixer_nid, 0, conn, ARRAY_SIZE(conn) - 1); for (i = 0; i < nums; i++) { if (get_wcaps_type(get_wcaps(codec, conn[i])) == AC_WID_AUD_OUT) @@ -882,7 +882,7 @@ static int add_secret_dac_path(struct hda_codec *codec) conn[nums++] = nid; return snd_hda_override_conn_list(codec, spec->gen.mixer_nid, - nums, conn); + 0, nums, conn); } } return 0; @@ -1082,8 +1082,10 @@ static void fix_vt1802_connections(struct hda_codec *codec) static hda_nid_t conn_24[] = { 0x14, 0x1c }; static hda_nid_t conn_33[] = { 0x1c }; - snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24); - snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33); + snd_hda_override_conn_list(codec, 0x24, 0, ARRAY_SIZE(conn_24), + conn_24); + snd_hda_override_conn_list(codec, 0x33, 0, ARRAY_SIZE(conn_33), + conn_33); } /* patch for vt2002P */ -- 1.9.1

3 12

[alsa-devel] [PATCH] ASoC: nau8825: add crosstalk suppression feature
by John Hsu 17 Mar '16

17 Mar '16

Add crosstalk signal suppression function. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 403 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 77 ++++++++- 2 files changed, 479 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index d270f63..81554be 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -82,6 +82,380 @@ static const struct nau8825_fll_attr fll_pre_scalar[] = { { 8, 0x3 }, }; +/* Crosstalk detection */ +#define LOG10_MAGIC 646456993 +#define GAIN_AUGMENT 22500 +#define SIDETONE_BASE 207000 + +static struct reg_default nau8825_xtalk_baktab[] = { + { NAU8825_REG_FLL6, 0 }, + { NAU8825_REG_CLK_DIVIDER, 0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0 }, + { NAU8825_REG_HSVOL_CTRL, 0 }, + { NAU8825_REG_DACL_CTRL, 0 }, + { NAU8825_REG_DACR_CTRL, 0 }, + { NAU8825_REG_CLASSG_CTRL, 0 }, +}; + +static const struct reg_default nau8825_biq_reg[] = { + { NAU8825_REG_BIQ_CTRL, 0x0000 }, + { NAU8825_REG_BIQ_COF1, 0x009B }, + { NAU8825_REG_BIQ_COF2, 0x0006 }, + { NAU8825_REG_BIQ_COF3, 0xff66 }, + { NAU8825_REG_BIQ_COF4, 0x0000 }, + { NAU8825_REG_BIQ_COF5, 0xffb3 }, + { NAU8825_REG_BIQ_COF6, 0x0000 }, + { NAU8825_REG_BIQ_COF7, 0x009A }, + { NAU8825_REG_BIQ_COF8, 0x0006 }, + { NAU8825_REG_BIQ_COF9, 0xffb3 }, + { NAU8825_REG_BIQ_COF10, 0x8000 }, + { NAU8825_REG_BIQ_CTRL, 0x0010 }, +}; + +static const unsigned short logtable[256] = { + 0x0000, 0x0171, 0x02e0, 0x044e, 0x05ba, 0x0725, 0x088e, 0x09f7, + 0x0b5d, 0x0cc3, 0x0e27, 0x0f8a, 0x10eb, 0x124b, 0x13aa, 0x1508, + 0x1664, 0x17bf, 0x1919, 0x1a71, 0x1bc8, 0x1d1e, 0x1e73, 0x1fc6, + 0x2119, 0x226a, 0x23ba, 0x2508, 0x2656, 0x27a2, 0x28ed, 0x2a37, + 0x2b80, 0x2cc8, 0x2e0f, 0x2f54, 0x3098, 0x31dc, 0x331e, 0x345f, + 0x359f, 0x36de, 0x381b, 0x3958, 0x3a94, 0x3bce, 0x3d08, 0x3e41, + 0x3f78, 0x40af, 0x41e4, 0x4319, 0x444c, 0x457f, 0x46b0, 0x47e1, + 0x4910, 0x4a3f, 0x4b6c, 0x4c99, 0x4dc5, 0x4eef, 0x5019, 0x5142, + 0x526a, 0x5391, 0x54b7, 0x55dc, 0x5700, 0x5824, 0x5946, 0x5a68, + 0x5b89, 0x5ca8, 0x5dc7, 0x5ee5, 0x6003, 0x611f, 0x623a, 0x6355, + 0x646f, 0x6588, 0x66a0, 0x67b7, 0x68ce, 0x69e4, 0x6af8, 0x6c0c, + 0x6d20, 0x6e32, 0x6f44, 0x7055, 0x7165, 0x7274, 0x7383, 0x7490, + 0x759d, 0x76aa, 0x77b5, 0x78c0, 0x79ca, 0x7ad3, 0x7bdb, 0x7ce3, + 0x7dea, 0x7ef0, 0x7ff6, 0x80fb, 0x81ff, 0x8302, 0x8405, 0x8507, + 0x8608, 0x8709, 0x8809, 0x8908, 0x8a06, 0x8b04, 0x8c01, 0x8cfe, + 0x8dfa, 0x8ef5, 0x8fef, 0x90e9, 0x91e2, 0x92db, 0x93d2, 0x94ca, + 0x95c0, 0x96b6, 0x97ab, 0x98a0, 0x9994, 0x9a87, 0x9b7a, 0x9c6c, + 0x9d5e, 0x9e4f, 0x9f3f, 0xa02e, 0xa11e, 0xa20c, 0xa2fa, 0xa3e7, + 0xa4d4, 0xa5c0, 0xa6ab, 0xa796, 0xa881, 0xa96a, 0xaa53, 0xab3c, + 0xac24, 0xad0c, 0xadf2, 0xaed9, 0xafbe, 0xb0a4, 0xb188, 0xb26c, + 0xb350, 0xb433, 0xb515, 0xb5f7, 0xb6d9, 0xb7ba, 0xb89a, 0xb97a, + 0xba59, 0xbb38, 0xbc16, 0xbcf4, 0xbdd1, 0xbead, 0xbf8a, 0xc065, + 0xc140, 0xc21b, 0xc2f5, 0xc3cf, 0xc4a8, 0xc580, 0xc658, 0xc730, + 0xc807, 0xc8de, 0xc9b4, 0xca8a, 0xcb5f, 0xcc34, 0xcd08, 0xcddc, + 0xceaf, 0xcf82, 0xd054, 0xd126, 0xd1f7, 0xd2c8, 0xd399, 0xd469, + 0xd538, 0xd607, 0xd6d6, 0xd7a4, 0xd872, 0xd93f, 0xda0c, 0xdad9, + 0xdba5, 0xdc70, 0xdd3b, 0xde06, 0xded0, 0xdf9a, 0xe063, 0xe12c, + 0xe1f5, 0xe2bd, 0xe385, 0xe44c, 0xe513, 0xe5d9, 0xe69f, 0xe765, + 0xe82a, 0xe8ef, 0xe9b3, 0xea77, 0xeb3b, 0xebfe, 0xecc1, 0xed83, + 0xee45, 0xef06, 0xefc8, 0xf088, 0xf149, 0xf209, 0xf2c8, 0xf387, + 0xf446, 0xf505, 0xf5c3, 0xf680, 0xf73e, 0xf7fb, 0xf8b7, 0xf973, + 0xfa2f, 0xfaea, 0xfba5, 0xfc60, 0xfd1a, 0xfdd4, 0xfe8e, 0xff47 +}; + +/** + * computes log10 of a value; the result is round off to 3 decimal. + * this function takes reference to dvb-math. + * + * @return log10(value) * 1000 + */ +static u32 nau8825_intlog10_dec3(u32 value) +{ + u32 msb, logentry, significand, interpolation, log10val; + u64 log2val; + + msb = fls(value) - 1; + significand = value << (31 - msb); + logentry = (significand >> 23) & 0xff; + interpolation = ((significand & 0x7fffff) * + ((logtable[(logentry + 1) & 0xff] - + logtable[logentry]) & 0xffff)) >> 15; + + log2val = ((msb << 24) + (logtable[logentry] << 8) + interpolation); + log10val = (log2val * LOG10_MAGIC) >> 31; + return log10val / ((1 << 24) / 1000); +} + +/** + * computes cross-talk sidetone with orignal and cross-talk signal + * + * @sig_org: orignal signal + * @sig_cros: cross-talk signal + * + * return cross-talk sidetone + */ +static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) +{ + u32 gain, sidetone; + + if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { + WARN_ON(1); + return 0; + } + + sig_org = nau8825_intlog10_dec3(sig_org); + sig_cros = nau8825_intlog10_dec3(sig_cros); + gain = (sig_org - sig_cros) * 20 + GAIN_AUGMENT; + sidetone = SIDETONE_BASE - gain * 2; + sidetone /= 1000; + + return sidetone; +} + +static void nau8825_xtalk_backup(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) + regmap_read(nau8825->regmap, nau8825_xtalk_baktab[i].reg, + &nau8825_xtalk_baktab[i].def); +} + +static void nau8825_xtalk_restore(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) + regmap_write(nau8825->regmap, nau8825_xtalk_baktab[i].reg, + nau8825_xtalk_baktab[i].def); +} + +static void nau8825_xtalk_biq_apply(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_biq_reg[i].reg, + nau8825_biq_reg[i].def); +} + +static void nau8825_xtalk_biq_cancel(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_biq_reg[i].reg, 0); +} + +static void nau8825_xtalk_prepare_dac(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDACL_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, NAU8825_HP_BOOST_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR | NAU8825_ENABLE_DACL | + NAU8825_ENABLE_ADC | NAU8825_ENABLE_ADC_CLK | + NAU8825_ENABLE_DAC_CLK, NAU8825_ENABLE_DACR | + NAU8825_ENABLE_DACL | NAU8825_ENABLE_ADC | + NAU8825_ENABLE_ADC_CLK | NAU8825_ENABLE_DAC_CLK); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW | NAU8825_CHANRGE_PUMP_EN, + NAU8825_JAMNODCLOW | NAU8825_CHANRGE_PUMP_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L | + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L | + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, 0); +} + +static void nau8825_xtalk_prepare_adc(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ANALOG_ADC_2, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_MASK, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB); +} + +static void nau8825_xtalk_clock(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, NAU8825_DCO_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_VCO | 0x3); +} + +static void nau8825_xtalk_prepare(struct nau8825 *nau8825) +{ + /* backup value of specific register used by cross-talk */ + nau8825_xtalk_backup(nau8825); + nau8825_xtalk_clock(nau8825); + nau8825_xtalk_prepare_dac(nau8825); + nau8825_xtalk_prepare_adc(nau8825); + /* Config channel */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_MASK, NAU8825_DACR_CH_SEL_R); + /* Config headphone volume */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HPL_VOL_MASK | NAU8825_HPR_VOL_MASK, 0x0492); + /* Config IIS parameter */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_DRV_MASK | NAU8825_I2S_BLK_DIV_MASK, + (0x2 << NAU8825_I2S_DRV_SFT) | 0x1); + /* Config crosstalk */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, + NAU8825_IMM_THD_MASK | NAU8825_IMM_CYC_MASK | + NAU8825_IMM_DAC_SRC_MASK, (0x2 << NAU8825_IMM_THD_SFT) | + NAU8825_IMM_CYC_8192 | NAU8825_IMM_DAC_SRC_SIN); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_INTERRUPT_MASK, NAU8825_IRQ_RMS_EN, 0); + nau8825_xtalk_biq_apply(nau8825); +} + +static void nau8825_xtalk_clean_dac(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, NAU8825_HP_BOOST_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL | + NAU8825_JAMNODCLOW); + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_CHANRGE_PUMP_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP | + NAU8825_BIAS_TESTDAC_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL | NAU8825_ENABLE_ADC | + NAU8825_ENABLE_ADC_CLK | NAU8825_ENABLE_DAC_CLK, 0); + if (!nau8825->irq) + regmap_update_bits(nau8825->regmap, + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN, 0); +} + +static void nau8825_xtalk_clean_adc(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ANALOG_ADC_2, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_MASK, 0); +} + +static void nau8825_xtalk_clean(struct nau8825 *nau8825) +{ + /* restore value of specific register for cross-talk */ + nau8825_xtalk_restore(nau8825); + nau8825_xtalk_clean_dac(nau8825); + nau8825_xtalk_clean_adc(nau8825); + regmap_write(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_RMS_EN, NAU8825_IRQ_RMS_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + nau8825_xtalk_biq_cancel(nau8825); +} + +static void nau8825_xtalk_imm_start(struct nau8825 *nau8825, int vol) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ADC_DGAIN_CTRL, + NAU8825_ADC_DIG_VOL_MASK, vol); + switch (nau8825->xtalk_state) { + case NAU8825_XTALK_HPR_R2L: + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP, + NAU8825_BIAS_HPR_IMP); + break; + case NAU8825_XTALK_HPL_R2L: + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP, + NAU8825_BIAS_HPL_IMP); + break; + default: + break; + } + msleep(100); + regmap_update_bits(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, + NAU8825_IMM_EN, NAU8825_IMM_EN); +} + +static void nau8825_xtalk_imm_stop(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_IMM_MODE_CTRL, NAU8825_IMM_EN, 0); +} + +static void nau8825_xtalk_measure(struct nau8825 *nau8825) +{ + u32 sidetone; + + switch (nau8825->xtalk_state) { + case NAU8825_XTALK_DONE: + nau8825_xtalk_prepare(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_HPR_R2L; + msleep(280); + nau8825_xtalk_imm_start(nau8825, 0x00d2); + break; + case NAU8825_XTALK_HPR_R2L: + regmap_read(nau8825->regmap, NAU8825_REG_IMM_RMS_L, + &nau8825->imp_rms[NAU8825_XTALK_HPR_R2L]); + dev_dbg(nau8825->dev, "HPR_R2L imm: %x\n", + nau8825->imp_rms[NAU8825_XTALK_HPR_R2L]); + nau8825_xtalk_imm_stop(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_HPL_R2L; + nau8825_xtalk_imm_start(nau8825, 0x00ff); + break; + case NAU8825_XTALK_HPL_R2L: + regmap_read(nau8825->regmap, NAU8825_REG_IMM_RMS_L, + &nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + dev_dbg(nau8825->dev, "HPL_R2L imm: %x\n", + nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + nau8825_xtalk_imm_stop(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_IMM; + break; + case NAU8825_XTALK_IMM: + sidetone = nau8825_xtalk_sidetone( + nau8825->imp_rms[NAU8825_XTALK_HPR_R2L], + nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + dev_dbg(nau8825->dev, "cross-talk sidetone: %x\n", sidetone); + regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, + (sidetone << 8) | sidetone); + nau8825_xtalk_clean(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_DONE; + break; + default: + break; + } +} + +static void nau8825_xtalk_work(struct work_struct *work) +{ + struct nau8825 *nau8825 = container_of( + work, struct nau8825, xtalk_work); + + nau8825_xtalk_measure(nau8825); + if (nau8825->xtalk_state == NAU8825_XTALK_IMM) + nau8825_xtalk_measure(nau8825); +} + + static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_ENA_CTRL, 0x00ff }, { NAU8825_REG_IIC_ADDR_SET, 0x0 }, @@ -211,6 +585,7 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) case NAU8825_REG_RESET: case NAU8825_REG_IRQ_STATUS: case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_BIQ_CTRL ... NAU8825_REG_BIQ_COF10: case NAU8825_REG_IMM_RMS_L: case NAU8825_REG_IMM_RMS_R: case NAU8825_REG_I2C_DEVICE_ID: @@ -741,6 +1116,10 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); mic_detected = (jack_status_reg >> 10) & 3; + if (mic_detected == 0x3) + nau8825->high_imped = 1; + else + nau8825->high_imped = 0; switch (mic_detected) { case 0: @@ -836,6 +1215,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); + if (!nau8825->high_imped) { + nau8825->xtalk_state = NAU8825_XTALK_DONE; + schedule_work(&nau8825->xtalk_work); + } } else { dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); nau8825_eject_jack(nau8825); @@ -843,6 +1226,9 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { + schedule_work(&nau8825->xtalk_work); + clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; } if (!clear_irq) @@ -901,6 +1287,11 @@ static void nau8825_init_regs(struct nau8825 *nau8825) /* Latch IIC LSB value */ regmap_write(regmap, NAU8825_REG_IIC_ADDR_SET, 0x0001); + /* Config channel */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_MASK, NAU8825_DACR_CH_SEL_R); /* Enable Bias/Vmid */ regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); @@ -1007,6 +1398,17 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); nau8825_restart_jack_detection(nau8825->regmap); + INIT_WORK(&nau8825->xtalk_work, nau8825_xtalk_work); + + return 0; +} + +static int nau8825_codec_remove(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + cancel_work_sync(&nau8825->xtalk_work); + return 0; } @@ -1326,6 +1728,7 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, + .remove = nau8825_codec_remove, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 029134a..808474d 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -98,7 +98,13 @@ #define NAU8825_ENABLE_DACR_SFT 10 #define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) #define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_DACL (1 << NAU8825_ENABLE_DACL_SFT) #define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_ADC (1 << NAU8825_ENABLE_ADC_SFT) +#define NAU8825_ENABLE_ADC_CLK_SFT 7 +#define NAU8825_ENABLE_ADC_CLK (1 << NAU8825_ENABLE_ADC_CLK_SFT) +#define NAU8825_ENABLE_DAC_CLK_SFT 6 +#define NAU8825_ENABLE_DAC_CLK (1 << NAU8825_ENABLE_DAC_CLK_SFT) #define NAU8825_ENABLE_SAR_SFT 1 /* CLK_DIVIDER (0x3) */ @@ -153,6 +159,7 @@ /* INTERRUPT_MASK (0xf) */ #define NAU8825_IRQ_OUTPUT_EN (1 << 11) #define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_RMS_EN (1 << 8) #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) @@ -225,10 +232,13 @@ /* I2S_PCM_CTRL2 (0x1d) */ #define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ +#define NAU8825_I2S_DRV_SFT 12 +#define NAU8825_I2S_DRV_MASK (0x3 << NAU8825_I2S_DRV_SFT) #define NAU8825_I2S_MS_SFT 3 #define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_BLK_DIV_MASK 0x7 /* ADC_RATE (0x2b) */ #define NAU8825_ADC_SYNC_DOWN_SFT 0 @@ -247,22 +257,66 @@ #define NAU8825_DAC_OVERSAMPLE_128 2 #define NAU8825_DAC_OVERSAMPLE_32 4 +/* ADC_DGAIN_CTRL (0x30) */ +#define NAU8825_ADC_DAC_ST1_SFT 12 +#define NAU8825_ADC_DAC_ST1_MASK 0xf +#define NAU8825_ADC_DAC_ST2_SFT 8 +#define NAU8825_ADC_DAC_ST2_MASK 0xf +#define NAU8825_ADC_DIG_VOL_MASK 0xff + /* MUTE_CTRL (0x31) */ #define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) #define NAU8825_DAC_SOFT_MUTE (1 << 9) /* HSVOL_CTRL (0x32) */ #define NAU8825_HP_MUTE (1 << 15) +#define NAU8825_HP_MUTE_AUTO (1 << 14) +#define NAU8825_HPL_MUTE (1 << 13) +#define NAU8825_HPR_MUTE (1 << 12) +#define NAU8825_HPL_VOL_SFT 6 +#define NAU8825_HPL_VOL_MASK (0x3f << NAU8825_HPL_VOL_SFT) +#define NAU8825_HPR_VOL_SFT 0 +#define NAU8825_HPR_VOL_MASK (0x3f << NAU8825_HPR_VOL_SFT) /* DACL_CTRL (0x33) */ #define NAU8825_DACL_CH_SEL_SFT 9 +#define NAU8825_DACL_CH_SEL_MASK (0x1 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_L (0x0 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_R (0x1 << NAU8825_DACL_CH_SEL_SFT) /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +#define NAU8825_DACR_CH_SEL_MASK (0x1 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_L (0x0 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_R (0x1 << NAU8825_DACR_CH_SEL_SFT) + +/* IMM_MODE_CTRL (0x4C) */ +#define NAU8825_IMM_THD_SFT 8 +#define NAU8825_IMM_THD_MASK (0x3f << NAU8825_IMM_THD_SFT) +#define NAU8825_IMM_CYC_SFT 4 +#define NAU8825_IMM_CYC_MASK (0x3 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_1024 (0x0 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_2048 (0x1 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_4096 (0x2 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_8192 (0x3 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_EN (1 << 3) +#define NAU8825_IMM_DAC_SRC_MASK 0x7 +#define NAU8825_IMM_DAC_SRC_BIQ 0x0 +#define NAU8825_IMM_DAC_SRC_DRC 0x1 +#define NAU8825_IMM_DAC_SRC_MIX 0x2 +#define NAU8825_IMM_DAC_SRC_SIN 0x3 /* CLASSG_CTRL (0x50) */ #define NAU8825_CLASSG_TIMER_SFT 8 #define NAU8825_CLASSG_TIMER_MASK (0x3f << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_1ms (0x1 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_2ms (0x2 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_8ms (0x4 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_16ms (0x8 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_32ms (0x10 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_64ms (0x20 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_LDAC_EN (0x1 << 2) +#define NAU8825_CLASSG_RDAC_EN (0x1 << 1) #define NAU8825_CLASSG_EN (1 << 0) /* I2C_DEVICE_ID (0x58) */ @@ -271,7 +325,12 @@ #define NAU8825_SOFTWARE_ID_NAU8825 0x0 /* BIAS_ADJ (0x66) */ -#define NAU8825_BIAS_TESTDAC_EN (0x3 << 8) +#define NAU8825_BIAS_HPR_IMP (1 << 15) +#define NAU8825_BIAS_HPL_IMP (1 << 14) +#define NAU8825_BIAS_TESTDAC_SFT 8 +#define NAU8825_BIAS_TESTDAC_EN (0x3 << NAU8825_BIAS_TESTDAC_SFT) +#define NAU8825_BIAS_TESTDACR_EN (0x2 << NAU8825_BIAS_TESTDAC_SFT) +#define NAU8825_BIAS_TESTDACL_EN (0x1 << NAU8825_BIAS_TESTDAC_SFT) #define NAU8825_BIAS_VMID (1 << 6) #define NAU8825_BIAS_VMID_SEL_SFT 4 #define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) @@ -290,6 +349,10 @@ #define NAU8825_POWERUP_ADCL (1 << 6) /* RDAC (0x73) */ +#define NAU8825_RDAC_EN_SFT 12 +#define NAU8825_RDAC_EN (0x3 << NAU8825_RDAC_EN_SFT) +#define NAU8825_RDAC_CLK_EN_SFT 8 +#define NAU8825_RDAC_CLK_EN (0x3 << NAU8825_RDAC_CLK_EN_SFT) #define NAU8825_RDAC_CLK_DELAY_SFT 4 #define NAU8825_RDAC_CLK_DELAY_MASK (0x7 << NAU8825_RDAC_CLK_DELAY_SFT) #define NAU8825_RDAC_VREF_SFT 2 @@ -333,12 +396,21 @@ enum { NAU8825_CLK_FLL_FS, }; +/* Crosstalk detection state */ +enum { + NAU8825_XTALK_HPR_R2L = 0, + NAU8825_XTALK_HPL_R2L, + NAU8825_XTALK_IMM, + NAU8825_XTALK_DONE, +}; + struct nau8825 { struct device *dev; struct regmap *regmap; struct snd_soc_dapm_context *dapm; struct snd_soc_jack *jack; struct clk *mclk; + struct work_struct xtalk_work; int irq; int mclk_freq; /* 0 - mclk is disabled */ int button_pressed; @@ -357,6 +429,9 @@ struct nau8825 { int key_debounce; int jack_insert_debounce; int jack_eject_debounce; + int xtalk_state; + int imp_rms[NAU8825_XTALK_IMM]; + int high_imped; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

2 4