February 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] Trivial fixes in INSTALL file
by Vikram Narayanan 13 Feb '12

13 Feb '12

Changed from HG->git in the INSTALL file Signed-off-by: Vikram Narayanan <vikram186(a)gmail.com> --- diff --git a/INSTALL b/INSTALL index 202c491..544d5d2 100644 --- a/INSTALL +++ b/INSTALL @@ -24,11 +24,11 @@ package. The configure script from alsa-utils package probably cannot find header file asoundlib.h in $prefix/include/alsa directory (usually in /usr/include/alsa directory). -Compilation from HG sources +Compilation from git sources --------------------------- You need also GNU packages autoconf and automake installed in your system -to compile HG (Mercurial) sources of alsa-utils package. +to compile git sources of alsa-utils package. For compilation you can use these commands: @@ -40,7 +40,7 @@ For compilation you can use these commands: ./configure make -The included hgcompile script does this job for you. +The included gitcompile script does this job for you. Note: Some automake packages have missing aclocal program. Use newer version in the case. -- 1.7.4.1

2 2

Re: [alsa-devel] VT1705 : Mic not working with power_save
by Anisse Astier 13 Feb '12

13 Feb '12

(adding back list in Cc, it was lost) On Mon, 13 Feb 2012 10:02:27 +0800, <LydiaWang(a)viatech.com.cn> wrote : > > -----邮件原件----- > > 发件人: Anisse Astier [mailto:anisse@astier.eu] > > 发送时间: 2012年2月11日 1:56 > > 收件人: alsa-devel(a)alsa-project.org > > 抄送: Takashi Iwai; Lydia Wang > > 主题: VT1705 : Mic not working with power_save > > > > Hi, > > > > On the hardware described in attached alsa-info (VT1705), I have the > > following problem: > > - mic doesn't work > > - if, while recording, we switch "input source", mic works during the > > recording session. > > > > Passing power_save=0 as a parameter to snd-hda-intel solves the problem. > > > > It was reproduced on linus' master branch, tiwai's for-next branch, > > and 3.2 branch. > > > > Regards, > > > > Anisse > > Hi, > > The log showed that your machine had two microphones, one is an internal mic, > and the other is a rear mic. Which microphone did you use? The log showed > currently used mic is the internal mic. The rear mic was not plugged in a jack. > > Lydia On this hardware, there's only an internal mic, and no physical jack to plug another input.

3 5

[alsa-devel] Adding sound to BeagleBone
by Peter Bolch 13 Feb '12

13 Feb '12

Hi, what - beside making the hardware - has to be done in principle to add an external audio codec to BeagleBone board (OMAPAM3358 based, with I2S but without a codec on board)? To make things easy I will use a codec with available driver. Regards Peter

2 1

[alsa-devel] [PATCH] ASoC: wm9081: Use module_i2c_driver
by Mark Brown 12 Feb '12

12 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 23 +---------------------- 1 files changed, 1 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 13bfcbd..076c126 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1418,28 +1418,7 @@ static struct i2c_driver wm9081_i2c_driver = { }; #endif -static int __init wm9081_modinit(void) -{ - int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&wm9081_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(wm9081_modinit); - -static void __exit wm9081_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm9081_i2c_driver); -#endif -} -module_exit(wm9081_exit); - +module_i2c_driver(wm9081_i2c_driver); MODULE_DESCRIPTION("ASoC WM9081 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm9081: Move WM9081 IRQ platform data handling into I2C probe
by Mark Brown 12 Feb '12

12 Feb '12

Better style and better supports idle_bias_off which we're going to implement. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 20 +++++++++----------- 1 files changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 7b09b1f..5f464ab 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1258,7 +1258,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); int ret; - u16 reg; codec->control_data = wm9081->regmap; @@ -1268,16 +1267,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) return ret; } - reg = 0; - if (wm9081->pdata.irq_high) - reg |= WM9081_IRQ_POL; - if (!wm9081->pdata.irq_cmos) - reg |= WM9081_IRQ_OP_CTRL; - snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL, - WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); - - wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Enable zero cross by default */ snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, WM9081_LINEOUTZC, WM9081_LINEOUTZC); @@ -1395,6 +1384,15 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), sizeof(wm9081->pdata)); + reg = 0; + if (wm9081->pdata.irq_high) + reg |= WM9081_IRQ_POL; + if (!wm9081->pdata.irq_cmos) + reg |= WM9081_IRQ_OP_CTRL; + regmap_update_bits(wm9081->regmap, WM9081_INTERRUPT_CONTROL, + WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) -- 1.7.9.rc1

1 1

[alsa-devel] [PATCH 1/2] ASoC: wm8988: Convert to direct regmap API usage
by Mark Brown 12 Feb '12

12 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8988.c | 140 ++++++++++++++++++++++++++++++++++++++------- 1 files changed, 120 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 4ef9d4c..2470321 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -33,24 +33,89 @@ * We can't read the WM8988 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8988_reg[] = { - 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ - 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ - 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ - 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ - 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ - 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ - 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ - 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ - 0x0079, 0x0079, 0x0079, /* 40 */ +static const struct reg_default wm8988_reg_defaults[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x0079 }, + { 3, 0x0079 }, + { 5, 0x0008 }, + { 7, 0x000a }, + { 8, 0x0000 }, + { 10, 0x00ff }, + { 11, 0x00ff }, + { 12, 0x000f }, + { 13, 0x000f }, + { 16, 0x0000 }, + { 17, 0x007b }, + { 18, 0x0000 }, + { 19, 0x0032 }, + { 20, 0x0000 }, + { 21, 0x00c3 }, + { 22, 0x00c3 }, + { 23, 0x00c0 }, + { 24, 0x0000 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0050 }, + { 35, 0x0050 }, + { 36, 0x0050 }, + { 37, 0x0050 }, + { 40, 0x0079 }, + { 41, 0x0079 }, + { 42, 0x0079 }, }; +static bool wm8988_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8988_LINVOL: + case WM8988_RINVOL: + case WM8988_LOUT1V: + case WM8988_ROUT1V: + case WM8988_ADCDAC: + case WM8988_IFACE: + case WM8988_SRATE: + case WM8988_LDAC: + case WM8988_RDAC: + case WM8988_BASS: + case WM8988_TREBLE: + case WM8988_RESET: + case WM8988_3D: + case WM8988_ALC1: + case WM8988_ALC2: + case WM8988_ALC3: + case WM8988_NGATE: + case WM8988_LADC: + case WM8988_RADC: + case WM8988_ADCTL1: + case WM8988_ADCTL2: + case WM8988_PWR1: + case WM8988_PWR2: + case WM8988_ADCTL3: + case WM8988_ADCIN: + case WM8988_LADCIN: + case WM8988_RADCIN: + case WM8988_LOUTM1: + case WM8988_LOUTM2: + case WM8988_ROUTM1: + case WM8988_ROUTM2: + case WM8988_LOUT2V: + case WM8988_ROUT2V: + case WM8988_LPPB: + return true; + default: + return false; + } +} + /* codec private data */ struct wm8988_priv { + struct regmap *regmap; unsigned int sysclk; - enum snd_soc_control_type control_type; struct snd_pcm_hw_constraint_list *sysclk_constraints; }; @@ -661,6 +726,7 @@ static int wm8988_mute(struct snd_soc_dai *dai, int mute) static int wm8988_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 pwr_reg = snd_soc_read(codec, WM8988_PWR1) & ~0x1c1; switch (level) { @@ -674,7 +740,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8988->regmap); /* VREF, VMID=2x5k */ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); @@ -730,7 +796,10 @@ static struct snd_soc_dai_driver wm8988_dai = { static int wm8988_suspend(struct snd_soc_codec *codec) { + struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); + wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_mark_dirty(wm8988->regmap); return 0; } @@ -745,7 +814,8 @@ static int wm8988_probe(struct snd_soc_codec *codec) struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type); + codec->control_data = wm8988->regmap; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -781,9 +851,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .suspend = wm8988_suspend, .resume = wm8988_resume, .set_bias_level = wm8988_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8988_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8988_reg, .controls = wm8988_snd_controls, .num_controls = ARRAY_SIZE(wm8988_snd_controls), @@ -793,6 +860,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .num_dapm_routes = ARRAY_SIZE(wm8988_dapm_routes), }; +static struct regmap_config wm8988_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8988_LPPB, + .writeable_reg = wm8988_writeable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8988_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8988_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8988_spi_probe(struct spi_device *spi) { @@ -804,18 +883,28 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi) if (wm8988 == NULL) return -ENOMEM; - wm8988->control_type = SND_SOC_SPI; + wm8988->regmap = regmap_init_spi(spi, &wm8988_regmap); + if (IS_ERR(wm8988->regmap)) { + ret = PTR_ERR(wm8988->regmap); + dev_err(&spi->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8988); ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); + if (ret != 0) + regmap_exit(wm8988->regmap); return ret; } static int __devexit wm8988_spi_remove(struct spi_device *spi) { + struct wm8988_priv *wm8988 = spi_get_drvdata(spi); snd_soc_unregister_codec(&spi->dev); + regmap_exit(wm8988->regmap); return 0; } @@ -842,16 +931,27 @@ static __devinit int wm8988_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8988); - wm8988->control_type = SND_SOC_I2C; + + wm8988->regmap = regmap_init_i2c(i2c, &wm8988_regmap); + if (IS_ERR(wm8988->regmap)) { + ret = PTR_ERR(wm8988->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); + if (ret != 0) + regmap_exit(wm8988->regmap); + return ret; } static __devexit int wm8988_i2c_remove(struct i2c_client *client) { + struct wm8988_priv *wm8988 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + regmap_exit(wm8988->regmap); return 0; } -- 1.7.9.rc1

1 1

[alsa-devel] [PATCH 1/4] ASoC: wm_hubs: Improve single ended line output enable performance
by Mark Brown 12 Feb '12

12 Feb '12

The enable of the single ended line outputs on wm_hubs devices performs better if the output is enabled prior to starting VMID. Since inactive outputs are held at VMID anyway there is little cost to doing this for unused outputs. Add callbacks into wm_hubs and keep track of which outputs are really active so we can disable them once we're active. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 4 ++ sound/soc/codecs/wm8994.c | 5 ++ sound/soc/codecs/wm_hubs.c | 113 ++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_hubs.h | 11 ++++ 4 files changed, 125 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index ab4685a..1e69f63 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1058,6 +1058,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; + wm_hubs_set_bias_level(codec, level); + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -1078,6 +1080,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(wm8993->regmap, false); regcache_sync(wm8993->regmap); + wm_hubs_vmid_ena(codec); + /* Bring up VMID with fast soft start */ snd_soc_update_bits(codec, WM8993_ANTIPOP2, WM8993_STARTUP_BIAS_ENA | diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e6eebf7..21931a0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -787,6 +787,8 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | (0x3 << WM8994_VMID_RAMP_SHIFT)); + wm_hubs_vmid_ena(codec); + /* Remove discharge for line out */ snd_soc_update_bits(codec, WM8994_ANTIPOP_1, WM8994_LINEOUT1_DISCH | @@ -2074,6 +2076,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + wm_hubs_set_bias_level(codec, level); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -2168,6 +2172,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, wm8994->cur_fw = NULL; break; } + codec->dapm.bias_level = level; return 0; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a53daf6..f7650c5 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -500,6 +500,36 @@ static int earpiece_event(struct snd_soc_dapm_widget *w, return 0; } +static int lineout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + bool *flag; + + switch (w->shift) { + case WM8993_LINEOUT1N_ENA_SHIFT: + flag = &hubs->lineout1n_ena; + break; + case WM8993_LINEOUT1P_ENA_SHIFT: + flag = &hubs->lineout1p_ena; + break; + case WM8993_LINEOUT2N_ENA_SHIFT: + flag = &hubs->lineout2n_ena; + break; + case WM8993_LINEOUT2P_ENA_SHIFT: + flag = &hubs->lineout2p_ena; + break; + default: + WARN(1, "Unknown line output"); + return -EINVAL; + } + + *flag = SND_SOC_DAPM_EVENT_ON(event); + + return 0; +} + static const struct snd_kcontrol_new in1l_pga[] = { SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0), SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0), @@ -675,14 +705,18 @@ SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0, line2p_mix, ARRAY_SIZE(line2p_mix)), -SND_SOC_DAPM_OUT_DRV("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUTPUT("SPKOUTLP"), SND_SOC_DAPM_OUTPUT("SPKOUTLN"), @@ -949,6 +983,11 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int jd_scthr, int jd_thr, int micbias1_lvl, int micbias2_lvl) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + hubs->lineout1_se = !lineout1_diff; + hubs->lineout2_se = !lineout2_diff; + if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, WM8993_LINEOUT1_MODE, @@ -978,6 +1017,64 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata); +void wm_hubs_vmid_ena(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int val = 0; + + if (hubs->lineout1_se) + val |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + val |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; + + /* Enable the line outputs while we power up */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, val, val); +} +EXPORT_SYMBOL_GPL(wm_hubs_vmid_ena); + +void wm_hubs_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int val; + + switch (level) { + case SND_SOC_BIAS_ON: + /* Turn off any unneded single ended outputs */ + val = 0; + + if (hubs->lineout1_se && hubs->lineout1n_ena) + val |= WM8993_LINEOUT1N_ENA; + + if (hubs->lineout1_se && hubs->lineout1p_ena) + val |= WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se && hubs->lineout2n_ena) + val |= WM8993_LINEOUT2N_ENA; + + if (hubs->lineout2_se && hubs->lineout2p_ena) + val |= WM8993_LINEOUT2P_ENA; + + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, + WM8993_LINEOUT1N_ENA | + WM8993_LINEOUT1P_ENA | + WM8993_LINEOUT2N_ENA | + WM8993_LINEOUT2P_ENA, + val); + + if (!hubs->lineout1n_ena && !hubs->lineout1p_ena && + !hubs->lineout2n_ena && !hubs->lineout2p_ena) + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, 0); + break; + + default: + break; + } +} +EXPORT_SYMBOL_GPL(wm_hubs_set_bias_level); + MODULE_DESCRIPTION("Shared support for Wolfson hubs products"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index c674c7a..4140905 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -33,6 +33,14 @@ struct wm_hubs_data { bool class_w; u16 class_w_dcs; + bool lineout1_se; + bool lineout1n_ena; + bool lineout1p_ena; + + bool lineout2_se; + bool lineout2n_ena; + bool lineout2p_ena; + bool dcs_done_irq; struct completion dcs_done; }; @@ -46,5 +54,8 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); +extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec); +extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); #endif -- 1.7.9.rc1

1 3

[alsa-devel] [PATCH v2 00/12] ASoC: ALC5632: Capture path improvements
by Leon Romanovsky 12 Feb '12

12 Feb '12

This patches is bringing improved support of capture path to ALC5632 codec. --- Sorry for resending it, I rebased first patchset on the wrong branch. Andrey Danin (10): ASoC: alc5632: Coding style. Remove two extra empty lines. ASoC: alc5632: Fixed voice DAC volume step. ASoC: alc5632: Add voice DAC playback switch ASoC: alc5632: Rename capture switches to common scheme XXX2REC ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix ASoC: alc5632: Refactored DAPM routes to add voice support ASoC: alc5632: Add DMIC switches and controls ASoC: alc5632: Fix I2S digital interface power for recording ASoC: alc5632: Allow 8kHz stream support. ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix Paul Fertser (2): ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs sound/soc/codecs/alc5632.c | 183 +++++++++++++++++++++++++++++++------------- sound/soc/codecs/alc5632.h | 1 + 2 files changed, 130 insertions(+), 54 deletions(-) -- 1.7.3.4

2 18

[alsa-devel] [PATCH 00/12] ASoC: ALC5632: Capture path improvements
by Leon Romanovsky 11 Feb '12

11 Feb '12

This patches is bringing improved support of capture path to ALC5632 codec. Andrey Danin (10): ASoC: alc5632: Coding style. Remove two extra empty lines. ASoC: alc5632: Fixed voice DAC volume step. ASoC: alc5632: Add voice DAC playback switch ASoC: alc5632: Rename capture switches to common scheme XXX2REC ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix ASoC: alc5632: Refactored DAPM routes to add voice support ASoC: alc5632: Add DMIC switches and controls ASoC: alc5632: Fix I2S digital interface power for recording ASoC: alc5632: Allow 8kHz stream support. ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix Paul Fertser (2): ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs sound/soc/codecs/alc5632.c | 183 +++++++++++++++++++++++++++++++------------- sound/soc/codecs/alc5632.h | 1 + 2 files changed, 130 insertions(+), 54 deletions(-) -- 1.7.3.4

1 12

[alsa-devel] ALSA 1.0.25 broke AC3 passthrough?
by Jean-Yves Avenard 11 Feb '12

11 Feb '12

Hi Today, I installed 1.0.25 drivers and libraries... AC3 passthrough is completely broken. My hardware: NVidia GT430, hdmi output Device used: hdmi:CARD=NVidia,DEV=0,AES0=6,AES1=0x82,AES2=0x00,AES3=0x01 on Ubuntu 10.04 LTS (but using manually built alsa) It makes the application hangs or behaves incorrectly. All the time broken, no matter which app. Reverting back to 1.0.24 fixes the problem I haven't delved much into it yet, MythTV (I am the audio developer) after opening the audio device in passthrough mode, trying to write audio on the pcm just hangs and result in an I/O error after a little while snd_pcm_writei returns an error (not immediately) snd_pcm_state(pcm_handle) returns 3, snd_strerror(err) returns "Input/output error" mplayer just hangs... nothing obvious in the logs, audio device opens okay, i can't say I notice much difference in opening the device, allocating fragment and the lot. Anyone else has seen this? Jean-Yves

2 4