September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: HDA: Add support for IDT 92HD93
by David Henningsson 20 Sep '11

20 Sep '11

Cc: stable(a)kernel.org BugLink: http://bugs.launchpad.net/bugs/854468 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 5145b66..7d8efab 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -6573,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx}, -- 1.7.5.4

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[alsa-devel] [PATCH 1/1] ymfpci - enable rear device and dup4ch control when ac97 support 4 channels
by Raymond Yau 20 Sep '11

20 Sep '11

ymfpci - enable rear device and dup4ch control when ac97 support 4 channels

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Re: [alsa-devel] hda- Disable Sticky PCM
by Takashi Iwai 20 Sep '11

20 Sep '11

At Tue, 20 Sep 2011 08:15:09 +0800, Raymond Yau wrote: > > 2011/9/18 Takashi Iwai <tiwai(a)suse.de>: > > > > Well, I can't reproduce it in hda-emu, so I don't know now. > > Try to get a stacktrace to see what's calling there actually. > > > > > set stream active : active stream 1 > ALSA hda_intel.c:1757: azx_pcm_prepare: bufsize=0xd750, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x4, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x3, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x6, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x5, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0xa, stream=0x5, > channel=0, format=0x4111 > hda-intel: IRQ timing workaround is activated for card #1. Suggest a > bigger bdl_pos_adj. > ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x4 > ------------[ cut here ]------------ (snip) Thanks, so it's called from hw_free. This is a call just to be sure, and should be harmless. If you mind, the verbs are suppressed by a patch like below. Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6b611d5..e3db196 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1491,8 +1491,11 @@ static void really_cleanup_stream(struct hda_codec *codec, struct hda_cvt_setup *q) { hda_nid_t nid = q->nid; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); + if (q->stream_tag || q->channel_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + if (q->format_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0 +); memset(q, 0, sizeof(*q)); q->nid = nid; }

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[alsa-devel] [PATCH] ASoC: Fix unused variable warning in WM8996
by Mark Brown 19 Sep '11

19 Sep '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f455997..bede1ac 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -670,8 +670,6 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, static int bg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int ret = 0; switch (event) { -- 1.7.6.3

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[PATCH 2/2] [ASoC] 88pm860x-codec - reset the codec correctly
by Bas Vermeulen 19 Sep '11

19 Sep '11

Reset the codec according to the Audio power-up delay errata for the 88PM8607. Signed-off-by: Bas Vermeulen <bas.vermeulen(a)novero.com> --- sound/soc/codecs/88pm860x-codec.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 0198dbb..df7b4a0 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1179,6 +1179,9 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, REG_MISC2, data); + udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; pm860x_reg_write(codec->control_data, REG_MISC2, data); -- 1.7.0.4 --------------080107070107020807050104--

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[PATCH 1/2] [ASoC] 88pm860x-codec - Allow independent use of both I2S playback and capture
by Bas Vermeulen 19 Sep '11

19 Sep '11

Introduce a I2S CLK supply so playback and capture can operate independently. Signed-off-by: Bas Vermeulen <bas.vermeulen(a)novero.com> --- sound/soc/codecs/88pm860x-codec.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 1924157..0198dbb 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -772,11 +772,12 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), @@ -868,6 +869,11 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left ADC", NULL, "Left ADC MOD"}, {"Right ADC", NULL, "Right ADC MOD"}, + /* I2S Clock */ + {"I2S DIN", NULL, "I2S CLK"}, + {"I2S DIN1", NULL, "I2S CLK"}, + {"I2S DOUT", NULL, "I2S CLK"}, + /* PCM/AIF1 Inputs */ {"PCM SDO", NULL, "ADC Left Mux"}, {"PCM SDO", NULL, "ADCR EC Mux"}, -- 1.7.0.4 --------------030808060208020409080204--

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Re: [alsa-devel] [PATCH] ASoC: sn95031: Staticize sn95031_pcm_hw_params
by Mark Brown 19 Sep '11

19 Sep '11

On Mon, Sep 19, 2011 at 04:34:28PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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[alsa-devel] [RFC PATCH] Fixup automute for Realtek auto parser
by David Henningsson 19 Sep '11

19 Sep '11

The ~6 months old functionality for configurable automuting is broken for the case that the user has only HP and LO (no speakers) - basically there is an "Automute mode" control, but enabling it does nothing. While fixing that, I also took the liberty of refactoring/rewriting some of that code, in order to make it easier to understand and maintain. Takashi, given you approve/like these changes, I'll go ahead and fix up the quirk files to fit the new variable names. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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Re: [alsa-devel] [PATCH] ASoC: sn95031: Staticize sn95031_pcm_hw_params
by Vinod Koul 19 Sep '11

19 Sep '11

On Mon, 2011-09-19 at 16:34 +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/codecs/sn95031.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c > index b4f1cb4..29945b0 100644 > --- a/sound/soc/codecs/sn95031.c > +++ b/sound/soc/codecs/sn95031.c > @@ -658,7 +658,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) > return 0; > } > > -int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, > +static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, > struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) > { > unsigned int format, rate; Acked-by: Vinod Koul <vinod.koul(a)linux.intel.com> -- ~Vinod

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[alsa-devel] [PATCH 1/2] ASoC: Initial WM8996 headphone impedance measurement support
by Mark Brown 19 Sep '11

19 Sep '11

The WM8996 can measure the impedance of accessories connected to the headphone output. Implement initial support for this, measuring the left channel impedance when an accessory is detected and using this to distinguish between a line load and a headphone load. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 137 ++++++++++++++++++++++++++++++++++++-------- 1 files changed, 112 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8e9d7f7..a6168c5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2350,12 +2350,94 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* Enable interrupts and we're off */ snd_soc_update_bits(codec, WM8996_INTERRUPT_STATUS_2_MASK, - WM8996_IM_MICD_EINT, 0); + WM8996_IM_MICD_EINT | WM8996_HP_DONE_EINT, 0); return 0; } EXPORT_SYMBOL_GPL(wm8996_detect); +static void wm8996_hpdet_irq(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int val, reg, report; + + /* Assume headphone in error conditions; we need to report + * something or we stall our state machine. + */ + report = SND_JACK_HEADPHONE; + + reg = snd_soc_read(codec, WM8996_HEADPHONE_DETECT_2); + if (reg < 0) { + dev_err(codec->dev, "Failed to read HPDET status\n"); + goto out; + } + + if (!(reg & WM8996_HP_DONE)) { + dev_err(codec->dev, "Got HPDET IRQ but HPDET is busy\n"); + goto out; + } + + val = reg & WM8996_HP_LVL_MASK; + + dev_dbg(codec->dev, "HPDET measured %d ohms\n", val); + + /* If we've got high enough impedence then report as line, + * otherwise assume headphone. + */ + if (val >= 126) + report = SND_JACK_LINEOUT; + else + report = SND_JACK_HEADPHONE; + +out: + if (wm8996->jack_mic) + report |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(wm8996->jack, report, + SND_JACK_LINEOUT | SND_JACK_HEADSET); + + wm8996->detecting = false; + + /* If the output isn't running re-clamp it */ + if (!(snd_soc_read(codec, WM8996_POWER_MANAGEMENT_1) & + (WM8996_HPOUT1L_ENA | WM8996_HPOUT1R_RMV_SHORT))) + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, 0); + + /* Go back to looking at the microphone */ + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, + WM8996_MICD_ENA); + + snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); +} + +static void wm8996_hpdet_start(struct snd_soc_codec *codec) +{ + /* Unclamp the output, we can't measure while we're shorting it */ + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT); + + /* We need bandgap for HPDET */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); + + /* Go into headphone detect left mode */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0); + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 1); + + /* Trigger a measurement */ + snd_soc_update_bits(codec, WM8996_HEADPHONE_DETECT_1, + WM8996_HP_POLL, WM8996_HP_POLL); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2376,28 +2458,36 @@ static void wm8996_micd(struct snd_soc_codec *codec) wm8996->jack_mic = false; wm8996->detecting = true; snd_soc_jack_report(wm8996->jack, 0, - SND_JACK_HEADSET | SND_JACK_BTN_0); + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, WM8996_MICD_RATE_MASK); return; } - /* If the measurement is very high we've got a microphone but - * do a little debounce to account for mechanical issues. + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. */ if (val & 0x400) { - dev_dbg(codec->dev, "Microphone detected\n"); - snd_soc_jack_report(wm8996->jack, SND_JACK_HEADSET, - SND_JACK_HEADSET | SND_JACK_BTN_0); - wm8996->jack_mic = true; - wm8996->detecting = false; - - /* Increase poll rate to give better responsiveness - * for buttons */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + if (wm8996->detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm8996->jack_mic = true; + wm8996_hpdet_start(codec); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK, + 5 << WM8996_MICD_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); + } + + return; } /* If we detected a lower impedence during initial startup @@ -2429,15 +2519,11 @@ static void wm8996_micd(struct snd_soc_codec *codec) if (val & 0x3fc) { if (wm8996->jack_mic) { dev_dbg(codec->dev, "Mic button detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADSET | SND_JACK_BTN_0, - SND_JACK_HEADSET | SND_JACK_BTN_0); - } else { - dev_dbg(codec->dev, "Headphone detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADPHONE, - SND_JACK_HEADSET | + snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); + } else if (wm8996->detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); /* Increase the detection rate a bit for * responsiveness. @@ -2445,8 +2531,6 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, 7 << WM8996_MICD_RATE_SHIFT); - - wm8996->detecting = false; } } } @@ -2486,6 +2570,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); + if (irq_val & WM8996_HP_DONE_EINT) + wm8996_hpdet_irq(codec); + return IRQ_HANDLED; } -- 1.7.6.3

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