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

[alsa-devel] [PATCH 1/2 2.6.38] ASoC: soc-cache: Introduce the cache_bypass option
by Dimitris Papastamos 11 Feb '11

11 Feb '11

This is primarily needed to avoid writing back to the cache whenever we are syncing the cache with the hardware. This gives a performance benefit especially for large register maps. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 36 ++++++++++++++++++++++++------------ 2 files changed, 25 insertions(+), 12 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 74921f2..5e19644 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -463,6 +463,7 @@ struct snd_soc_codec { /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; + unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int cache_only:1; /* Suppress writes to hardware */ unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ unsigned int suspended:1; /* Codec is in suspend PM state */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8c2a21a..799e376 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -25,7 +25,8 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -49,7 +50,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -106,7 +108,8 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -130,7 +133,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -191,7 +195,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -216,7 +221,8 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -271,7 +277,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -295,7 +302,8 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -450,7 +458,8 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -476,7 +485,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -568,7 +578,8 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -595,7 +606,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[3] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; -- 1.7.4

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Re: [alsa-devel] [PATCH] at73c213: add support for at73c240 devices
by Grant Likely 11 Feb '11

11 Feb '11

On Fri, Feb 11, 2011 at 8:56 AM, Andreas Bießmann <biessmann(a)corscience.de> wrote: > at73c240 is a successor of mature at73c213 and nearly register compatible. > > See http://www.atmel.com/dyn/resources/prod_documents/doc6484.pdf for > comparison of these two devices. > > Signed-off-by: Andreas Bießmann <biessmann(a)corscience.de> Looks okay to me. Briefly-glanced-at-by: Grant Likely <grant.likely(a)secretlab.ca> > --- > include/linux/spi/at73c213.h | 1 + > sound/spi/at73c213.c | 122 +++++++++++++++++++++++++++++++++--------- > sound/spi/at73c213.h | 3 + > 3 files changed, 101 insertions(+), 25 deletions(-) > > diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h > index 0f20a70e..26d1f38 100644 > --- a/include/linux/spi/at73c213.h > +++ b/include/linux/spi/at73c213.h > @@ -19,6 +19,7 @@ > struct at73c213_board_info { > int ssc_id; > struct clk *dac_clk; > + bool is_at73c240; > char shortname[32]; > }; > > diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c > index 503ffb0..711aae4 100644 > --- a/sound/spi/at73c213.c > +++ b/sound/spi/at73c213.c > @@ -709,10 +709,44 @@ AT73C213_MONO_SWITCH("Aux Capture Switch", 0, DAC_CTRL, DAC_CTRL_ONAUXIN, > AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), > }; > > +static struct snd_kcontrol_new snd_at73c240_controls[] __devinitdata = { > +AT73C213_STEREO("Master Playback Volume", 0, DAC_LMPG, DAC_RMPG, 0, 0, 0x1f, 1), > +AT73C213_STEREO("Master Playback Switch", 0, DAC_LMPG, DAC_RMPG, 5, 5, 1, 1), > +AT73C213_STEREO("PCM Playback Volume", 0, DAC_LLOG, DAC_RLOG, 0, 0, 0x1f, 1), > +AT73C213_STEREO("PCM Playback Switch", 0, DAC_LLOG, DAC_RLOG, 5, 5, 1, 1), > +AT73C213_MONO_SWITCH("Mono PA Playback Switch", 0, DAC_CTRL, > + (DAC_CTRL_ONPADRV-1), 0x01, 0), > +{ > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "PA Playback Volume", > + .index = 0, > + .info = snd_at73c213_pa_volume_info, > + .get = snd_at73c213_mono_get, > + .put = snd_at73c213_mono_put, > + .private_value = PA_CTRL | (PA_CTRL_APAGAIN << 8) | \ > + (0x0f << 16) | (1 << 24), > +}, > +AT73C213_MONO_SWITCH("PA Playback Switch", 0, PA_CTRL, (PA_CTRL_APAON-1), > + 0x01, 0), > +{ > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Line Capture Volume", > + .index = 0, > + .info = snd_at73c213_line_capture_volume_info, > + .get = snd_at73c213_stereo_get, > + .put = snd_at73c213_stereo_put, > + .private_value = DAC_LLIG | (DAC_RLIG << 8) | (0 << 16) | (0 << 19) > + | (0x1f << 24) | (1 << 22), > +}, > +AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), > +}; > + > static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) > { > struct snd_card *card; > int errval, idx; > + struct snd_kcontrol_new *controls; > + int controls_size = 0; > > if (chip == NULL || chip->pcm == NULL) > return -EINVAL; > @@ -721,9 +755,17 @@ static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) > > strcpy(card->mixername, chip->pcm->name); > > - for (idx = 0; idx < ARRAY_SIZE(snd_at73c213_controls); idx++) { > + if (chip->board->is_at73c240) { > + controls_size = ARRAY_SIZE(snd_at73c240_controls); > + controls = snd_at73c240_controls; > + } else { > + controls_size = ARRAY_SIZE(snd_at73c213_controls); > + controls = snd_at73c213_controls; > + } > + > + for (idx = 0; idx < controls_size; idx++) { > errval = snd_ctl_add(card, > - snd_ctl_new1(&snd_at73c213_controls[idx], > + snd_ctl_new1(&controls[idx], > chip)); > if (errval < 0) > goto cleanup; > @@ -732,7 +774,7 @@ static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) > return 0; > > cleanup: > - for (idx = 1; idx < ARRAY_SIZE(snd_at73c213_controls) + 1; idx++) { > + for (idx = 1; idx < controls_size + 1; idx++) { > struct snd_kcontrol *kctl; > kctl = snd_ctl_find_numid(card, idx); > if (kctl) > @@ -777,7 +819,7 @@ static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) > static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > { > int retval; > - unsigned char dac_ctrl = 0; > + unsigned char tmp_reg; > > retval = snd_at73c213_set_bitrate(chip); > if (retval) > @@ -799,7 +841,11 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0xff); > if (retval) > goto out_clk; > - retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APAPRECH)); > + if (chip->board->is_at73c240) > + tmp_reg = (1<<(PA_CTRL_APAPRECH-1)); > + else > + tmp_reg = (1<<PA_CTRL_APAPRECH); > + retval = snd_at73c213_write_reg(chip, PA_CTRL, tmp_reg); > if (retval) > goto out_clk; > retval = snd_at73c213_write_reg(chip, DAC_CTRL, > @@ -809,11 +855,13 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > > msleep(50); > > - /* Stop precharging PA. */ > - retval = snd_at73c213_write_reg(chip, PA_CTRL, > - (1<<PA_CTRL_APALP) | 0x0f); > - if (retval) > - goto out_clk; > + if (!chip->board->is_at73c240) { > + /* Stop precharging PA. */ > + retval = snd_at73c213_write_reg(chip, PA_CTRL, > + (1<<PA_CTRL_APALP) | 0x0f); > + if (retval) > + goto out_clk; > + } > > msleep(450); > > @@ -825,12 +873,18 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > msleep(1); > > /* Turn on DAC. */ > - dac_ctrl = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) > - | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR) > - /* enable MONO PA */ > - | (1<<DAC_CTRL_ONPADRV); > + if (chip->board->is_at73c240) > + tmp_reg = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) > + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR) > + /* enable MONO PA */ > + | (1<<(DAC_CTRL_ONPADRV-1)); > + else > + tmp_reg = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) > + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR) > + /* enable MONO PA */ > + | (1<<DAC_CTRL_ONPADRV); > > - retval = snd_at73c213_write_reg(chip, DAC_CTRL, dac_ctrl); > + retval = snd_at73c213_write_reg(chip, DAC_CTRL, tmp_reg); > if (retval) > goto out_clk; > > @@ -859,9 +913,19 @@ static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) > retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); > if (retval) > goto out_clk; > - retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > - if (retval) > - goto out_clk; > + if (!chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > + if (retval) > + goto out_clk; > + } > + > + /* enable I2S clock */ > + if (chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, DAC_IS_CTRL, > + (1<<DAC_IS_CTRL_IS_CTRL)); > + if (retval) > + goto out_clk; > + } > > /* Enable I2S device, i.e. clock output. */ > ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); > @@ -915,6 +979,10 @@ static int __devinit snd_at73c213_dev_init(struct snd_card *card, > memcpy(&chip->reg_image, &snd_at73c213_original_image, > sizeof(snd_at73c213_original_image)); > > + /* at73c240 has another reset value here */ > + if (chip->board->is_at73c240) > + chip->reg_image[PA_CTRL] = 0x0f; > + > retval = snd_at73c213_ssc_init(chip); > if (retval) > goto out_irq; > @@ -1043,9 +1111,11 @@ static int __devexit snd_at73c213_remove(struct spi_device *spi) > retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); > if (retval) > goto out; > - retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > - if (retval) > - goto out; > + if (!chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); > + if (retval) > + goto out; > + } > > /* Turn off PA. */ > retval = snd_at73c213_write_reg(chip, PA_CTRL, > @@ -1053,10 +1123,12 @@ static int __devexit snd_at73c213_remove(struct spi_device *spi) > if (retval) > goto out; > msleep(10); > - retval = snd_at73c213_write_reg(chip, PA_CTRL, > - (1 << PA_CTRL_APALP) | 0x0f); > - if (retval) > - goto out; > + if (!chip->board->is_at73c240) { > + retval = snd_at73c213_write_reg(chip, PA_CTRL, > + (1 << PA_CTRL_APALP) | 0x0f); > + if (retval) > + goto out; > + } > > /* Turn off external DAC. */ > retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x0c); > diff --git a/sound/spi/at73c213.h b/sound/spi/at73c213.h > index fd8b372..3712d6e 100644 > --- a/sound/spi/at73c213.h > +++ b/sound/spi/at73c213.h > @@ -88,6 +88,9 @@ > #define DAC_MISC_DEEMPEN 2 > #define DAC_MISC_NBITS 0 > > +#define DAC_IS_CTRL 0x0B > +#define DAC_IS_CTRL_IS_CTRL 2 > + > /* DAC Precharge Control Register */ > #define DAC_PRECH 0x0C > #define DAC_PRECH_PRCHGPDRV 7 > -- > 1.7.2.3 > > > ------------------------------------------------------------------------------ > The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: > Pinpoint memory and threading errors before they happen. > Find and fix more than 250 security defects in the development cycle. > Locate bottlenecks in serial and parallel code that limit performance. > http://p.sf.net/sfu/intel-dev2devfeb > _______________________________________________ > spi-devel-general mailing list > spi-devel-general(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/spi-devel-general > -- Grant Likely, B.Sc., P.Eng. Secret Lab Technologies Ltd.

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[alsa-devel] finally, ALSA 1.0.24 is out
by Jaroslav Kysela 11 Feb '11

11 Feb '11

Hello all, after very intensive testing, I released all ALSA packages with version number 1.0.24(+) : * alsa-driver-1.0.24 * alsa-lib-1.0.24.1 * alsa-utils-1.0.24.2 * alsa-tools-1.0.24 * alsa-firmware-1.0.24 * alsa-plugins-1.0.24 * pyalsa-1.0.24 The full list of changes is here: http://www.alsa-project.org/main/index.php/Changes_v1.0.23_v1.0.24 There is a ton of work in these parts (among others): * alsa-driver - HDA driver - ASoC tree - USB driver * alsa-library - Use Case Manager * alsa-utils - alsaloop and alsaucm introduction - alsactl improvements * alsa-tools - updated hdspmixer Please, report compilation problems directly to me or to the alsa-devel mailing list. Enjoy this release, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] QasMixer 0.7.1 now available
by Sebastian H. 11 Feb '11

11 Feb '11

Hello Alsa and Qt users QasMixer version 0.7.1 is now available. This point release brings persistent application settings and some GUI cleanups. New features * Persistent application settings managed by QT's QSettings class * A settings dialog * Icons from the system theme are used for menus * A sort option for the Alsa configuration tree Changes * The Alsa configuration view has moved to a dialog * System tray command line options were removed in favour of application settings * Many small GUI tweaks * Some bug fixes The homepage has been updated with screenshots http://xwmw.org/qasmixer and files are hosted at sourceforge http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

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[alsa-devel] [PATCH 1/2 2.6.38] ASoC: WM8994: Improve robustness in some use cases
by Dimitris Papastamos 11 Feb '11

11 Feb '11

Ensure that on disabling certain registers such as AIF1DAC1L, AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly when enabling those registers, AIF1CLK and AIF2CLK will remain disabled. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 142 ++++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 133 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 37b8aa8..bd0cfdd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,9 @@ struct wm8994_priv { int revision; struct wm8994_pdata *pdata; + + unsigned int aif1clk_enable:1; + unsigned int aif2clk_enable:1; }; static int wm8994_readable(unsigned int reg) @@ -1004,6 +1007,82 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + if (wm8994->aif2clk_enable) + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + break; + } + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + wm8994->aif2clk_enable = 0; + } + break; + } + + return 0; +} + +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif1clk_enable = 1; + break; + } + + return 0; +} + +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif2clk_enable = 1; + break; + } + + return 0; +} + static const char *hp_mux_text[] = { "Mixer", "DAC", @@ -1272,6 +1351,29 @@ static const struct soc_enum aif2dacr_src_enum = static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); +static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) +}; + +static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1284,9 +1386,6 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, @@ -1516,14 +1615,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, /* DAC1 inputs */ - { "DAC1L", NULL, "DAC1L Mixer" }, { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, - { "DAC1R", NULL, "DAC1R Mixer" }, { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1532,7 +1629,6 @@ static const struct snd_soc_dapm_route intercon[] = { /* DAC2/AIF2 outputs */ { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, - { "DAC2L", NULL, "AIF2DAC2L Mixer" }, { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, @@ -1540,7 +1636,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, - { "DAC2R", NULL, "AIF2DAC2R Mixer" }, { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1584,6 +1679,24 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = { + { "DAC1L", NULL, "Late DAC1L Enable PGA" }, + { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "Late DAC1R Enable PGA" }, + { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "Late DAC2L Enable PGA" }, + { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "Late DAC2R Enable PGA" }, + { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" } +}; + +static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = { + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, +}; + static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF1DACDAT", NULL, "AIF2DACDAT" }, { "AIF2DACDAT", NULL, "AIF1DACDAT" }, @@ -3125,6 +3238,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); + if (wm8994->revision < 4) + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + else + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); break; case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, @@ -3143,10 +3262,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) + if (wm8994->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); - + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + } break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- 1.7.4

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Re: [alsa-devel] [PATCH 3/3] ASoC: Add support for AIF channel muxing on WM8903
by Stephen Warren 11 Feb '11

11 Feb '11

Stephen Warren wrote at Wednesday, February 09, 2011 9:33 AM: > > Mark Brown wrote at Wednesday, February 09, 2011 6:47 AM: > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > --- > > sound/soc/codecs/wm8903.c | 66 ++++++++++++++++++++++++++++++++++++++++++--- > > 1 files changed, 62 insertions(+), 4 deletions(-) > > ...I'll make sure to test it within the next few days, once it's applied. Mark, I did find one issue with this. With the default Left/Right Capture Mux settings, and the patch series I just posted, mic detection now works fine in all the scenarios I tested: 1) Mic plugged in before loading driver, or not. 2) Mic plugged in when starting to capture, or not. 3) Mic removed and plugged back in while capture active. However, if I flip the Right Capture Mux to setting Left (while not capturing), then this breaks case (3) above; the initial portion of the recording is fine, but when I remove and plug the mic back in, something isn't turned on, and so nothing is recorded after that. I haven't had a chance to look into this yet, but can later; probably on Saturday evening. -- nvpublic

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[alsa-devel] [PATCH] ASoC: Allow use sleeping gpio in soc-jack
by Jarkko Nikula 11 Feb '11

11 Feb '11

It is safe to use sleeping gpio in snd_soc_jack_gpio_detect as it is not called from interrupt context. This avoids WARN_ON from __gpio_get_value if sleeping gpio is registered for jack. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- Only compile tested. --- sound/soc/soc-jack.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 99dbaf7..4579ee0 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -240,7 +240,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) int enable; int report; - enable = gpio_get_value(gpio->gpio); + enable = gpio_get_value_cansleep(gpio->gpio); if (gpio->invert) enable = !enable; -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: mid-x86: Use the soc-jack apis for jack type detection
by Koul, Vinod 11 Feb '11

11 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch modifies the mfld_machine to use the new jack apis for adding the voltage zones for jack type detection. It also modifed TI sn95031 codec driver to use these new apis Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/codecs/sn95031.c | 4 +--- sound/soc/mid-x86/mfld_machine.c | 12 ++++++++++++ 2 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 4cc0017..d0b7802 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -782,9 +782,7 @@ static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) { int micbias = sn95031_get_mic_bias(mfld_jack->codec); - /* Defaulting to HEADSET for now. - * will change after adding soc-jack detection apis */ - int jack_type = SND_JACK_HEADSET; + int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 45a0067..96487fb 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -74,6 +74,12 @@ static struct snd_soc_jack_pin mfld_jack_pins[] = { }, }; +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, + {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -264,6 +270,12 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) pr_err("adding jack pins failed\n"); return ret_val; } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc -- 1.7.2.3

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[alsa-devel] [PATCH 1/2] ASoC: WM8994: Improve robustness in some use cases
by Dimitris Papastamos 11 Feb '11

11 Feb '11

Ensure that on disabling certain registers such as AIF1DAC1L, AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly when enabling those registers, AIF1CLK and AIF2CLK will remain disabled. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 142 ++++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 133 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0024b4c..055ad8f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,9 @@ struct wm8994_priv { int revision; struct wm8994_pdata *pdata; + + unsigned int aif1clk_enable:1; + unsigned int aif2clk_enable:1; }; static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) @@ -1004,6 +1007,82 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + if (wm8994->aif2clk_enable) + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + break; + } + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + wm8994->aif2clk_enable = 0; + } + break; + } + + return 0; +} + +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif1clk_enable = 1; + break; + } + + return 0; +} + +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8994->aif2clk_enable = 1; + break; + } + + return 0; +} + static const char *hp_mux_text[] = { "Mixer", "DAC", @@ -1272,6 +1351,29 @@ static const struct soc_enum aif2dacr_src_enum = static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); +static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) +}; + +static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1284,9 +1386,6 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, @@ -1516,14 +1615,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, /* DAC1 inputs */ - { "DAC1L", NULL, "DAC1L Mixer" }, { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, - { "DAC1R", NULL, "DAC1R Mixer" }, { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1532,7 +1629,6 @@ static const struct snd_soc_dapm_route intercon[] = { /* DAC2/AIF2 outputs */ { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, - { "DAC2L", NULL, "AIF2DAC2L Mixer" }, { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, @@ -1540,7 +1636,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, - { "DAC2R", NULL, "AIF2DAC2R Mixer" }, { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, @@ -1584,6 +1679,24 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = { + { "DAC1L", NULL, "Late DAC1L Enable PGA" }, + { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "Late DAC1R Enable PGA" }, + { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "Late DAC2L Enable PGA" }, + { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "Late DAC2R Enable PGA" }, + { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" } +}; + +static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = { + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, +}; + static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF1DACDAT", NULL, "AIF2DACDAT" }, { "AIF2DACDAT", NULL, "AIF1DACDAT" }, @@ -3125,6 +3238,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); + if (wm8994->revision < 4) + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + else + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); break; case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, @@ -3143,10 +3262,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) + if (wm8994->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); - + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + } break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- 1.7.3.5

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[alsa-devel] [PATCH 2.6.38] ASoC: CX20442: fix wrong reg_cache_default content
by Janusz Krzysztofik 11 Feb '11

11 Feb '11

Content of the CX20442's snd_soc_codec_driver.reg_cache_default pointed area, introduced with my recent NULL pointer dereferece fix (commit f019ee5feb344ff0b22b58df4568676295aae14f), occured wrong after further testing, more thorough than just booting successfully. There are two problems with it: 1) It should read (1 << CX20442_TELOUT) | (1 << CX20442_MIC), not CX20442_TELOUT | CX20442_MIC. 2) While correctly matching actual codec hardware state on boot when fixed per 1), a few more code modifications would still be required to reflect that state not only into register cache, but also force them into DAPM pins state, otherwise an inconsitency occures which may prevent further codec state changes from being applied correctly. As a result, the phone stops ringing after reboot, until someone picks up the handset for the first time. Revert that reg_cache_default content to a working, previous de facto default value of 0, in hope this change can still be accepted as an rc cycle fix. Created and tested against linux-2.6.38-rc4 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/codecs/cx20442.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- git/sound/soc/codecs/cx20442.c.orig 2011-02-10 00:52:33.000000000 +0100 +++ git/sound/soc/codecs/cx20442.c 2011-02-10 12:17:31.000000000 +0100 @@ -367,7 +367,7 @@ static int cx20442_codec_remove(struct s return 0; } -static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; +static const u8 cx20442_reg; static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe,

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