December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
by Alexander 17 Jan '11

17 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Changes to I2S code: - MCLK is turned on on driver probe. This is done for several codecs that need this for registers access (like CS4271). - SCLK and LRCLK rates are corrected, assuming we always send 32 bits * 2 channels to codec. - Formats list shortened to just S32_LE, this makes all the DMA transactions right, while ALSA will do all sample format translation for us. Changes to both I2S and PCM code: - Rates list extended up to 96kHz, it's tested on EDB9302 and works for both capture and playback. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 37 +++++++++++++++++++++++-------------- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 2 files changed, 25 insertions(+), 16 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 4f48733..e5e9b9a 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -98,7 +98,6 @@ static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream) if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { /* Enable clocks */ - clk_enable(info->mclk); clk_enable(info->sclk); clk_enable(info->lrclk); @@ -136,7 +135,6 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) /* Disable clocks */ clk_disable(info->lrclk); clk_disable(info->sclk); - clk_disable(info->mclk); } } @@ -267,14 +265,16 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len); /* - * Calculate the sdiv (bit clock) and lrdiv (left/right clock) values. - * If the lrclk is pulse length is larger than the word size, then the - * bit clock will be gated for the unused bits. + * EP93xx I2S module can be setup so SCLK / LRCLK value can be + * 32, 64, 128. MCLK / SCLK value can be 2 and 4. + * We set LRCLK equal to `rate' and minimum SCLK / LRCLK + * value is 64, because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. */ - div = (clk_get_rate(info->mclk) / params_rate(params)) * - params_channels(params); + div = clk_get_rate(info->mclk) / params_rate(params); for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1) + for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) if (sdiv * lrdiv == div) { found = 1; goto out; @@ -316,6 +316,7 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + clk_disable(info->mclk); } static int ep93xx_i2s_resume(struct snd_soc_dai *dai) @@ -325,6 +326,7 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) if (!dai->active) return; + clk_enable(info->mclk); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); } @@ -341,9 +343,7 @@ static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { .set_fmt = ep93xx_i2s_set_dai_fmt, }; -#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver ep93xx_i2s_dai = { .symmetric_rates= 1, @@ -352,13 +352,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_96000, .formats = EP93XX_I2S_FORMATS, }, .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_96000, .formats = EP93XX_I2S_FORMATS, }, .ops = &ep93xx_i2s_dai_ops, @@ -404,10 +404,16 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) goto fail_unmap_mem; } + /* Minimal MCLK freq to enable some codecs */ + err = clk_set_rate(info->mclk, 8000 * 64 * 4); + if (err) + goto fail_put_mclk; + clk_enable(info->mclk); + info->sclk = clk_get(&pdev->dev, "sclk"); if (IS_ERR(info->sclk)) { err = PTR_ERR(info->sclk); - goto fail_put_mclk; + goto fail_disable_mclk; } info->lrclk = clk_get(&pdev->dev, "lrclk"); @@ -426,6 +432,8 @@ fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: clk_put(info->sclk); +fail_disable_mclk: + clk_disable(info->mclk); fail_put_mclk: clk_put(info->mclk); fail_unmap_mem: @@ -444,6 +452,7 @@ static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); clk_put(info->lrclk); clk_put(info->sclk); + clk_disable(info->mclk); clk_put(info->mclk); iounmap(info->regs); release_mem_region(info->mem->start, resource_size(info->mem)); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 2f121dd..0667077 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_96000, .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_48000, + .rate_max = SNDRV_PCM_RATE_96000, .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |

3 17

[alsa-devel] [PATCH] ASoC: Avoid direct register cache access when setting defaults
by Mark Brown 17 Jan '11

17 Jan '11

Directly accessing the register cache means that we can't use anything except a flat register cache so use snd_soc_update_bits(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8523.c | 6 +++--- sound/soc/codecs/wm8741.c | 13 ++++++++----- sound/soc/codecs/wm8904.c | 41 ++++++++++++++++++++++++++--------------- sound/soc/codecs/wm8955.c | 27 +++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 30 ++++++++++++++++++++---------- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm9090.c | 41 ++++++++++++++++++++++++----------------- 7 files changed, 101 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5eb2f50..9b8603f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -414,7 +414,6 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret, i; codec->hw_write = (hw_write_t)i2c_master_send; @@ -471,8 +470,9 @@ static int wm8523_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU; - reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC; + snd_soc_update_bits(codec, WM8523_DAC_GAINR, + WM8523_DACR_VU, WM8523_DACR_VU); + snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC); wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 494f2d3..25af901 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -421,7 +421,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); @@ -437,10 +436,14 @@ static int wm8741_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU */ - reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL; - reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM; + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, ARRAY_SIZE(wm8741_snd_controls)); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9de44a4..86c1cb8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2436,19 +2436,28 @@ static int wm8904_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU; - reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU; - reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU | - WM8904_HPOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU | - WM8904_HPOUTRZC; - reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTRZC; - reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); /* Apply configuration from the platform data. */ if (wm8904->pdata) { @@ -2469,10 +2478,12 @@ static int wm8904_probe(struct snd_soc_codec *codec) /* Set Class W by default - this will be managed by the Class * G widget at runtime where bypass paths are available. */ - reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR; + snd_soc_update_bits(codec, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); /* Use normal bias source */ - reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL; + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 7167dfc..5e0214d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -934,16 +934,27 @@ static int wm8955_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU; - reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU; - reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC; - reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC; - reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC; - reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC; - reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC; + snd_soc_update_bits(codec, WM8955_LEFT_DAC_VOLUME, + WM8955_LDVU, WM8955_LDVU); + snd_soc_update_bits(codec, WM8955_RIGHT_DAC_VOLUME, + WM8955_RDVU, WM8955_RDVU); + snd_soc_update_bits(codec, WM8955_LOUT1_VOLUME, + WM8955_LO1VU | WM8955_LO1ZC, + WM8955_LO1VU | WM8955_LO1ZC); + snd_soc_update_bits(codec, WM8955_ROUT1_VOLUME, + WM8955_RO1VU | WM8955_RO1ZC, + WM8955_RO1VU | WM8955_RO1ZC); + snd_soc_update_bits(codec, WM8955_LOUT2_VOLUME, + WM8955_LO2VU | WM8955_LO2ZC, + WM8955_LO2VU | WM8955_LO2ZC); + snd_soc_update_bits(codec, WM8955_ROUT2_VOLUME, + WM8955_RO2VU | WM8955_RO2ZC, + WM8955_RO2VU | WM8955_RO2ZC); + snd_soc_update_bits(codec, WM8955_MONOOUT_VOLUME, + WM8955_MOZC, WM8955_MOZC); /* Also enable adaptive bass boost by default */ - reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB; + snd_soc_update_bits(codec, WM8955_BASS_CONTROL, WM8955_BB, WM8955_BB); /* Set platform data values */ if (pdata) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b9cb1fc..c827057 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3822,16 +3822,26 @@ static int wm8962_probe(struct snd_soc_codec *codec) } /* Latch volume update bits */ - reg_cache[WM8962_LEFT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_RIGHT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_LEFT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_RIGHT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_LEFT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_RIGHT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_SPKOUTL_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_SPKOUTR_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_HPOUTL_VOLUME] |= WM8962_HPOUT_VU; - reg_cache[WM8962_HPOUTR_VOLUME] |= WM8962_HPOUT_VU; + snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); wm8962_add_widgets(codec); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc344..30fb48e 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -965,7 +965,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) * written. */ for (i = 0; i < ARRAY_SIZE(update_reg); i++) - ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100); /* Reset the codec */ ret = snd_soc_write(codec, WM8978_RESET, 0); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a788c42..3c5d325 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -551,7 +551,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret; codec->control_data = wm9090->control_data; @@ -576,22 +575,30 @@ static int wm9090_probe(struct snd_soc_codec *codec) /* Configure some defaults; they will be written out when we * bring the bias up. */ - reg_cache[WM9090_IN1_LINE_INPUT_A_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1A_ZC; - reg_cache[WM9090_IN1_LINE_INPUT_B_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1B_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_A_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2A_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_B_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2B_ZC; - reg_cache[WM9090_SPEAKER_VOLUME_LEFT] |= - WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC; - reg_cache[WM9090_LEFT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC; - reg_cache[WM9090_RIGHT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC; - - reg_cache[WM9090_CLOCKING_1] |= WM9090_TOCLK_ENA; + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_A_VOLUME, + WM9090_IN1_VU | WM9090_IN1A_ZC, + WM9090_IN1_VU | WM9090_IN1A_ZC); + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_B_VOLUME, + WM9090_IN1_VU | WM9090_IN1B_ZC, + WM9090_IN1_VU | WM9090_IN1B_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_A_VOLUME, + WM9090_IN2_VU | WM9090_IN2A_ZC, + WM9090_IN2_VU | WM9090_IN2A_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_B_VOLUME, + WM9090_IN2_VU | WM9090_IN2B_ZC, + WM9090_IN2_VU | WM9090_IN2B_ZC); + snd_soc_update_bits(codec, WM9090_SPEAKER_VOLUME_LEFT, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC); + snd_soc_update_bits(codec, WM9090_LEFT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC); + snd_soc_update_bits(codec, WM9090_RIGHT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC); + + snd_soc_update_bits(codec, WM9090_CLOCKING_1, + WM9090_TOCLK_ENA, WM9090_TOCLK_ENA); wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- 1.7.2.3

2 1

[alsa-devel] [PATCH] powerpc/85xx: fix compatible properties of the P1022DS DMA nodes used for audio
by Timur Tabi 13 Jan '11

13 Jan '11

In order to prevent the fsl_dma driver from claiming the DMA channels that the P1022DS audio driver needs, the compatible properties for those nodes must say "fsl,ssi-dma-channel" instead of "fsl,eloplus-dma-channel". Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, please pick up this patch for 2.6.37. arch/powerpc/boot/dts/p1022ds.dts | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/powerpc/boot/dts/p1022ds.dts b/arch/powerpc/boot/dts/p1022ds.dts index 2bbecbb..69422eb 100644 --- a/arch/powerpc/boot/dts/p1022ds.dts +++ b/arch/powerpc/boot/dts/p1022ds.dts @@ -291,13 +291,13 @@ ranges = <0x0 0xc100 0x200>; cell-index = <1>; dma00: dma-channel@0 { - compatible = "fsl,eloplus-dma-channel"; + compatible = "fsl,ssi-dma-channel"; reg = <0x0 0x80>; cell-index = <0>; interrupts = <76 2>; }; dma01: dma-channel@80 { - compatible = "fsl,eloplus-dma-channel"; + compatible = "fsl,ssi-dma-channel"; reg = <0x80 0x80>; cell-index = <1>; interrupts = <77 2>; -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: tpa6130a2: Fix compiler warning
by Peter Ujfalusi 12 Jan '11

12 Jan '11

sound/soc/codecs/tpa6130a2.c: In function 'tpa6130a2_add_controls': sound/soc/codecs/tpa6130a2.c:342: warning: unused variable 'dapm' Introduced by commit 39646871a47fd8808c08de0ce7d7ca8393af2805 ("ASoC: tpa6130a2: Replace DAPM code with direct interface"). The DAPM code has been removed from the driver, but the dapm struct remained. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0a99f31..1f1ac81 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -339,7 +339,6 @@ EXPORT_SYMBOL_GPL(tpa6130a2_stereo_enable); int tpa6130a2_add_controls(struct snd_soc_codec *codec) { struct tpa6130a2_data *data; - struct snd_soc_dapm_context *dapm = &codec->dapm; if (tpa6130a2_client == NULL) return -ENODEV; -- 1.7.3.4

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[alsa-devel] [PATCH] ASoC: Fix resource leak if soc_register_ac97_dai_link failed
by Axel Lin 12 Jan '11

12 Jan '11

Properly free the resources in the case of soc_register_ac97_dai_link failure. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against Linus tree. Regards, Axel sound/soc/soc-core.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 441285a..d7f80c3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1619,12 +1619,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) #ifdef CONFIG_SND_SOC_AC97_BUS /* register any AC97 codecs */ for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); - goto probe_dai_err; - } + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + while (--i >= 0) + soc_unregister_ac97_dai_link(&card->rtd[i]); + goto probe_dai_err; } + } #endif card->instantiated = 1; -- 1.7.0.4

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[alsa-devel] HDA - hdmi_show_short_audio_desc puts wrong rates in eld#N
by Anssi Hannula 11 Jan '11

11 Jan '11

Hi! Just noticed that the sample rates reported in eld#X.Y are wrong. They are printed by hdmi_show_short_audio_desc() in hda_eld.c. That function uses the snd_print_pcm_rates() function to convert the rates bitmap to a string. However, snd_print_pcm_rates() takes a HDA rates bitmap, not a generic SNDRV_ one. (BTW, an interesting function name for an internal HDA thing...) This results in several errors. For example, my 32000 44100 48000 88200 96000 176400 192000 becomes 44100 48000 88200 176400 192000 384000. -- Anssi Hannula

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[alsa-devel] UCM updates and fixes
by Liam Girdwood 11 Jan '11

11 Jan '11

Jarolsav, This fixes the remaining issues I can currently see in the alsa-lib ucm branch. It also includes a patch to execute commands, although I dont know if this is what you had in mind for exec. I've also done quite a bit of testing now. What is your merge plan for UCM ? We really need UCM merged soon as there are many users who depend on it. Thanks Liam

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[alsa-devel] [RFC/PATCH v7 00/12] Media controller (core and V4L2)
by Laurent Pinchart 11 Jan '11

11 Jan '11

Hi everybody, Here is the seventh version of the media controller core and V4L2 patches. the first one to be sent outside of the linux-media mailing list. Quick reminder for those who missed the previous version. let me quote the documentation (Documentation/DocBook/v4l/media-controller.xml). "Discovering a [media] device internal topology, and configuring it at runtime, is one of the goals of the media controller API. To achieve this, hardware devices are modelled as an oriented graph of building blocks called entities connected through pads." The code has been extensively reviewed by the V4L community, and this version is the first one to incorporate comments from the ALSA community (big thanks to Mark Brown and Clemens Ladisch). Two issues are not fully addressed yet, namely power management (I need to discuss this some more with the ALSA developers to really understand their requirements) and entities type names. I'm still posting this for review, as other developers have showed interest in commenting on the code. I want to emphasize once again that the media controller API does not replace the V4L, DVB or ALSA APIs. It complements them. The first user of the media controller API is the OMAP3 ISP driver. You can find it (as well as these patches and other V4L-specific patches) in a git tree at http://git.linuxtv.org/pinchartl/media.git (media-0004-omap3isp branch). The OMAP3 ISP driver patches are regularly posted for review on the linux-media list. Laurent Pinchart (10): media: Media device node support media: Media device media: Entities, pads and links media: Media device information query media: Entities, pads and links enumeration media: Links setup media: Pipelines and media streams v4l: Add a media_device pointer to the v4l2_device structure v4l: Make video_device inherit from media_entity v4l: Make v4l2_subdev inherit from media_entity Sakari Ailus (2): media: Entity graph traversal media: Reference count and power handling Documentation/DocBook/media-entities.tmpl | 24 + Documentation/DocBook/media.tmpl | 3 + Documentation/DocBook/v4l/media-controller.xml | 89 +++ Documentation/DocBook/v4l/media-func-close.xml | 59 ++ Documentation/DocBook/v4l/media-func-ioctl.xml | 116 ++++ Documentation/DocBook/v4l/media-func-open.xml | 94 +++ .../DocBook/v4l/media-ioc-device-info.xml | 133 ++++ .../DocBook/v4l/media-ioc-enum-entities.xml | 308 +++++++++ Documentation/DocBook/v4l/media-ioc-enum-links.xml | 207 ++++++ Documentation/DocBook/v4l/media-ioc-setup-link.xml | 93 +++ Documentation/media-framework.txt | 383 +++++++++++ Documentation/video4linux/v4l2-framework.txt | 72 ++- drivers/media/Kconfig | 13 + drivers/media/Makefile | 10 +- drivers/media/media-device.c | 382 +++++++++++ drivers/media/media-devnode.c | 321 +++++++++ drivers/media/media-entity.c | 690 ++++++++++++++++++++ drivers/media/video/v4l2-dev.c | 49 ++- drivers/media/video/v4l2-device.c | 52 ++- drivers/media/video/v4l2-subdev.c | 41 ++- include/linux/Kbuild | 1 + include/linux/media.h | 132 ++++ include/media/media-device.h | 92 +++ include/media/media-devnode.h | 97 +++ include/media/media-entity.h | 148 +++++ include/media/v4l2-dev.h | 7 + include/media/v4l2-device.h | 4 + include/media/v4l2-subdev.h | 10 + 28 files changed, 3603 insertions(+), 27 deletions(-) create mode 100644 Documentation/DocBook/v4l/media-controller.xml create mode 100644 Documentation/DocBook/v4l/media-func-close.xml create mode 100644 Documentation/DocBook/v4l/media-func-ioctl.xml create mode 100644 Documentation/DocBook/v4l/media-func-open.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-device-info.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-entities.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-links.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-setup-link.xml create mode 100644 Documentation/media-framework.txt create mode 100644 drivers/media/media-device.c create mode 100644 drivers/media/media-devnode.c create mode 100644 drivers/media/media-entity.c create mode 100644 include/linux/media.h create mode 100644 include/media/media-device.h create mode 100644 include/media/media-devnode.h create mode 100644 include/media/media-entity.h -- Regards, Laurent Pinchart

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[alsa-devel] Bug in setting channels, related to recent ELD/EDID changes?
by VDR User 10 Jan '11

10 Jan '11

Hi. After updating my stable kernel from 2.6.35.7 to 2.6.36.2, I get a problem when tuning HDTV channels that use surround sound. The error I get is as follows: audio_alsa_out: Cannot set number of channels to 6 (err=-22:Invalid argument) audio_out: open failed! With the following xine error: ---------------------- (ERROR) ---------------------- The audio device is unavailable. Please verify if another program already uses it. ['Audio device unavailable' ''] ------------------ (END OF ERROR) ------------------- This problem did not exist in 2.6.35.7 alsa drivers (using snd-hda-intel). I have noticed that there has been recent changes associated with ELD/EDID that sets limitations on channels. First of all I think this is a horrible idea since many devices don't even contain correct EDID. It's simply too unreliable to use. Secondly, could the above problem be related to those changes? If anyone knows a fix for this or can point me in the right direction (aside of going back to 2.6.35.7), PLEASE fill me in. Here are some system specs: debian testing kernel snd-hda-intel driver used for hdmi audio on nvidia gt240 video card hdmi going to 7.1 surround sound receiver & setup Any help is greatly appreciated! Best regards, Derek

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[alsa-devel] Questions regarding ASoC
by Bensaid, Selma 07 Jan '11

07 Jan '11

Hi, I'm studying the transition of my ALSA Driver for TI WL1273 BT/FM into ASoC, and I have a question. The ALSA driver already developped provided one sound card with 2 PCM devices: - PCM devices: BT (capture + playback) - FM (capture). These 2 devices are exclusive. Q1- Within ASoC should I create 2 sounds card one for BT and one FM, or should I keep 1 sound card with 2 PCM devices? Q2- in my ALSA driver I have handled in the .open the exclusivity between the FM and BT, is there any way to handle this generically? Thanks for your answers. Selma. --------------------------------------------------------------------- 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.

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