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

[alsa-devel] Join play of pcm in multiple playback devices
by Irfan shaikh 03 Jan '12

03 Jan '12

Hello All, I have written an application which supports join play of PCM Audio. i.e Capture and write to playback device 1 and playback device 2 simultaneously. Below code is just to explain the above scenario : /######################/ while(1) { retval=snd_pcm_readi(pAlsaConfig->pcm_handle, buffer, pAlsaConfig->alsaBufferSize); snd_pcm_writei(pAlsaConfig->pcm_handle1, buffer,retval); snd_pcm_writei(pAlsaConfig->pcm_handle2, buffer,retval); } /######################/ In above case, renderer 2 lags behind the renderer 1 i.e there is delay between two audio devices when actually heard. Please help me if some better implementation for join play. I have seen some thing as snd_pcm_link(). But i was not clear with its usage, while doing snd_pcm_writei ??? Thanks and regards, Irfan

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[alsa-devel] [PATCH] ASoC: sta32x: Optimize the array work to find rate_min and rate_max
by Axel Lin 03 Jan '12

03 Jan '12

For a given ir and fs, there is at most one possible match for the case mclk_ratios[ir][j].ratio * fs == freq. Thus we can break from the inner loop once a match is found. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sta32x.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fbd1450..7db6fa5 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -522,6 +522,7 @@ static int sta32x_set_dai_sysclk(struct snd_soc_dai *codec_dai, rate_min = fs; if (fs > rate_max) rate_max = fs; + break; } } } -- 1.7.5.4

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Re: [alsa-devel] [PATCH] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 03 Jan '12

03 Jan '12

On Tue, Jan 03, 2012 at 05:35:26AM -0800, Reddy, MR Swami wrote: > ASoC: Add support for TI LM49453 Audio codec. So, overall there's quite a few issues here. Many of them are fairly minor but there's a few big ones. The biggest issue is that this code won't compile, you need to submit code against development versions of the kernel. There's also quite a few errors with the use of snd_soc_update_bits() that make me wonder how well tested this code is. > +/* Register default values for LM49453 driver. */ > +static const u8 lm49453_reg_defs[LM49453_MAX_REGISTER + 1] = { > + [0x60] = 0x01, > + [0x67] = 0x01, This (and pretty much all I2C/SPI devices) really should be using regmap directly - the register map is pretty sparse so it would benefit from the rbtree cache. > +struct lm49453_priv { > + enum snd_soc_control_type control_type; Not needed since you only support I2C. > +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { > +SOC_DAPM_SINGLE("Port1_1 HPL", LM49453_P0_DACHPL1_REG, 0, 1, 0), These should all have " Switch" at the end of the name so the UIs know how to handle them (similarly for all your other mixer controls). > +static const struct snd_kcontrol_new lm49453_snd_controls[] = { > + SOC_ENUM("Mic2Mode Capture Route", lm49453_mic2mode_enum), > + SOC_ENUM("DMIC12 Capture Route", lm49453_dmic12_cfg_enum), > + SOC_ENUM("DMIC34 Capture Route", lm49453_dmic34_cfg_enum), These three look like they should be DAPM? > + SOC_SINGLE_TLV("Mic1 Capture Volume", LM49453_P0_ADC_LEVELL_REG, > + 0, 6, 0, digital_tlv), > + SOC_SINGLE_TLV("Mic2 Capture Volume", LM49453_P0_ADC_LEVELR_REG, > + 0, 6, 0, digital_tlv), The register names suggest that these are really just left and right channels on the ADC, do they really only affect the microphone paths? > + SOC_SINGLE_TLV("PORT1_1_RX_LVL", LM49453_P0_PORT1_RX_LVL1_REG, > + 0, 3, 0, port_tlv), All these should have Volume at the end of their name so that UIs know how to display them. > + SOC_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, > + 0, 0x3F, 0, digital_tlv), These sidetone controls all look like they could be combined into stereo controls - is there a pressing reason why are they done as mono? If there is a comment would probably help. > + SND_SOC_DAPM_INPUT("PORT1_1_RX"), > + SND_SOC_DAPM_INPUT("PORT1_2_RX"), > + SND_SOC_DAPM_INPUT("PORT1_3_RX"), > + SND_SOC_DAPM_INPUT("PORT1_4_RX"), > + SND_SOC_DAPM_INPUT("PORT1_5_RX"), > + SND_SOC_DAPM_INPUT("PORT1_6_RX"), > + SND_SOC_DAPM_INPUT("PORT1_7_RX"), > + SND_SOC_DAPM_INPUT("PORT1_8_RX"), > + SND_SOC_DAPM_INPUT("PORT2_1_RX"), > + SND_SOC_DAPM_INPUT("PORT2_2_RX"), INPUT widgets should generally be line inputs... looking at the DAPM routes these are actually digital widgets floating inside the CODEC. It's not clear why they're represented though as they don't have any control associated with them. If you just need widgets to make the routing easier then use PGAs, Inputs get treated specially in power control and are likely to confuse things. > + SND_SOC_DAPM_MICBIAS("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0), > + SND_SOC_DAPM_MICBIAS("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0), Don't use MICBIAS widgets, use SUPPLY widgets - MICBIAS widgets will (eventually) be phased out as SUPPLY widgets do the same job but are easier to use in machine drivers. > + SND_SOC_DAPM_SUPPLY("DMIC1supply", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("DMIC2supply", SND_SOC_NOPM, 0, 0, NULL, 0), > + > + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("Earpiece Rail", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("Haptic Rail", SND_SOC_NOPM, 0, 0, NULL, 0), None of these widgets will do anything, they don't have any power bits or callbacks associated with them so won't do anything. Just delete them. > + /* Port1 TX controls */ > + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + > + /* Port2 TX controls */ > + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + > + SND_SOC_DAPM_MIXER("Port1 Playback", LM49453_P0_AUDIO_PORT1_BASIC_REG, > + 0, 0, NULL, 0), > + SND_SOC_DAPM_MIXER("Port2 Playback", LM49453_P0_AUDIO_PORT2_BASIC_REG, > + 0, 0, NULL, 0), Similar issues to the capture versions of these apply here. > + /* Sidetone controls */ > + SND_SOC_DAPM_OUT_DRV("Sidetone DAC HPL Playback", > + LM49453_P0_STN_SEL_REG, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("Sidetone DAC HPR Playback", > + LM49453_P0_STN_SEL_REG, 1, 0, NULL, 0), What exactly do these register bits do? The use of OUT_DRV widgets looks very suspicious for a sidetone, an output driver would usually be a power amplifier or transducer driver so I suspect the power sequencing will be all wrong. It looks like these are mixer input controls which wouldn't normally be widgets at all. > + { "Port1_8 Mixer", NULL, "PORT1_8_RX_LVL" }, > + { "Port1_8 Mixer", NULL, "ADC Left" }, > + { "Port1_8 Mixer", NULL, "ADC Right" }, > + { "Port1_8 Mixer", NULL, "DMIC1 Left" }, > + { "Port1_8 Mixer", NULL, "DMIC1 Right" }, > + { "Port1_8 Mixer", NULL, "DMIC2 Left" }, > + { "Port1_8 Mixer", NULL, "DMIC2 Right" }, All this looks pretty odd, do the mixers really have no control over their inputs and just mix every possible input in? > + { "AMIC1Bias", NULL, "AMIC1"}, > + { "Mic1_Input Capture Route", "MIC1", "AMIC1Bias"}, Your CODEC driver should not be connecting the microphone biases, the connections are board specific. > +/* PLL P Values array */ > +int pll_p[] = { 143, /* For 8 fs p val */ > + 123, /* For 11.2 fs p val*/ > + 71, /* For 16 fs p val*/ > + 61, /* For 22.1 fs p val*/ > + 47, /* For 24 fs p val*/ > + 35, /* For 32 fs p val*/ > + 30, /* For 44.1 fs p val*/ > + 23, /* For 48 fs p val*/ > + 23 /* For 96fs p val*/ > +}; > + switch (lm49453->fs_rate) { > + case 8000: > + pll_div->p1 = pll_p[0]; > + break; Eew, no. Why are you indexing into the table with magic numbers from a switch statement to assign constants? That's not not really legible. Either put the frequencies into the table and look up in the table or put the constants directly into the switch statement. > +static int lm49453_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, > + int source, unsigned int freq_in, > + unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); > + struct pll_state *state; > + struct _pll_div pll_div; > + unsigned int pwr_mask; > + int ret; > + > + memset(&pll_div, 0, sizeof(pll_div)); That memset is just going to mask warnings, it shouldn't be needed. > + /* Setting DAC clock dividers based on substream sample rate. */ > + switch (lm49453->fs_rate) { > + case 8000: > + case 16000: > + case 32000: > + case 24000: > + case 48000: > + val = 256; > + break; > + case 11025: > + case 22050: > + case 44100: > + val = 216; > + break; > + case 96000: > + val = 127; > + break; > + default: > + return -EINVAL; > + } val really isn't a meaningful variable name... > + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, val); > + snd_soc_write(codec, LM49453_P0_DAC_OT_CLK_DIV_REG, val); > + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, val); ...and frankly I've no idea what this code is doing, I'd expect a clock divider configuration to vary depending on the input clock as well as the sample rate. > + switch (lm49453->sysclk) { > + case 12288000: > + case 26000000: > + case 19000000: > + val = 0; > + break; > + case 48000: > + case 32576: > + val = 1; > + break; > + default: > + return -EINVAL; > + } > + > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), val); I don't think this does what you think it does, your mask and value don't overlap at all. Have you tested? The whole thing looks very suspicous, running audio directly off a 32.768kHz clock would be unusual. I rather suspect this ought not to be done in hw_params() at all but instead should be in set_sysclk(). > +static int lm49453_set_dai_clkdiv(struct snd_soc_dai *codec_dai, > + int div_id, int div) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + > + switch (div_id) { > + case LM49453_PMC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_PMC_CLK_DIV_REG, div); > + break; > + case LM49453_HSDET_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_HSDET_CLK_DIV_REG, div); > + break; > + case LM49453_DMIC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_DMIC_CLK_DIV_REG, div); > + break; > + case LM49453_ADC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, div); > + break; > + case LM49453_DAC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_DAC_OT_CLK_DIV_REG, div); > + break; Again, not overburned with documentation here but I rather suspect that the driver ought to be able to figure out how to configure most of these dividers by itself given the SYSCLK and current sample rate. > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_CBS_CFS: > + aif_val = !(LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT); > + break; Use ~ if that's what you mean, though frankly I'm just guessing. > + case SND_SOC_DAIFMT_CBS_CFM: > + aif_val = LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT; > + break; > + case SND_SOC_DAIFMT_CBM_CFS: > + aif_val = ~LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT; > + break; > + case SND_SOC_DAIFMT_CBM_CFM: > + aif_val = LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT; > + break; You appear to be setting the same setting the same register value for CBS_CFM as for CBM_CFM. Are you sure the device can support all these clocking modes? > + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, > + aif_val, aif_val); Another error with update_bits(). > +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, > + unsigned int freq, int dir) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); > + > + switch (freq) { > + case 12288000: /* Platform System Clock */ > + case 26000000: /* 26 MHz */ > + case 19200000: /* 19.2 MHz */ > + case 48000: /* 48 KHz */ > + case 32576: /* 32.576 KHz */ The first comment is specific to your board, the others are blindingly obvious - I'd suggest removing them all. > + lm49453->sysclk = freq; > + return 0; > + } > + return -EINVAL; Move this into the default: in the switch for legibility. > +static int lm49453_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *codec_dai) > +{ > + unsigned int startup_mask; > + struct snd_soc_codec *codec = codec_dai->codec; > + dev_dbg(codec->dev, "lm49453_startup\n"); > + > + startup_mask = LM49453_P0_PMC_SETUP_CHIP_EN | > + LM49453_P0_PMC_SETUP_PLL_EN; > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, startup_mask, > + startup_mask); CHIP_EN should be handled by DAPM or set_bias_level(), PLL_EN looks like it should be done by set_pll() but possibly also by DAPM. > +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) > +{ > + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(0), > + (!!mute << 0)); > + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1), > + (!!mute << 1)); Why are there two separate update bits operations here? > + case SND_SOC_BIAS_STANDBY: > + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { > + /* Power up with Headset active */ Why are you doing this? DAPM should be taking care of enabling outputs... > + reg = snd_soc_read(codec, LM49453_P0_PMC_SETUP_REG); > + reg &= ~(LM49453_P0_PMC_SETUP_CHIP_EN); > + snd_soc_write(codec, LM49453_P0_PMC_SETUP_REG, reg); > + } Use _update_bits(). > + case SND_SOC_BIAS_OFF: > + reg = snd_soc_read(codec, LM49453_P0_PMC_SETUP_REG); > + snd_soc_write(codec, LM49453_P0_PMC_SETUP_REG, > + (reg | LM49453_P0_PMC_SETUP_CHIP_EN)); _update_bits() if this should be here at all. > + break; > + } Indentation. > +int lm49453_headset_detected(struct snd_soc_codec *codec) > +{ > + u8 val = 0; > + val = snd_soc_read(codec, LM49453_P0_HSD_IRQ1_REG); > + return (val >> 6) & 1; > +} > +EXPORT_SYMBOL_GPL(lm49453_headset_detected); Don't invent your own API for accessory detection - use the standard mechanisms we've got for reporting events. > +static int lm49453_probe(struct snd_soc_codec *codec) > +{ > + int ret = 0; > + int reg; > + > + codec->dapm.idle_bias_off = 1; Set this in the snd_soc_codec_driver structure. > + .reg_word_size = sizeof(u8), > + .reg_cache_default = lm49453_reg_defs, > + .compress_type = SND_SOC_RBTREE_COMPRESSION, This won't compile with current kernels, advanced cache types are no longer supported in ASoC. You should use regmap directly. > +#define LM49453_CLKSRC_MCLK 1 > +#define LM49453_CLKSRC_PLL_1 2 > +#define LM49453_CLKSRC_PLL_2 3 > +#define LM49453_CLKSRC_PORT1_RX 4 > +#define LM49453_CLKSRC_PORT2_RX 5 > +#define LM49453_CLKSRC_DAC_HP 6 > +#define LM49453_CLKSRC_DAC_OT 7 At no point are any of these referenced in the code... > +#define LM49453_CLKSRC_ADC 12 > +#define LM49453_CLKSRC_DMIC1 13 > +#define LM49453_CLKSRC_DMIC2 14 Really? > +/* Clock divider Id's */ ' shouldn't be used for plurals.

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[alsa-devel] [PATCH v2] ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs
by Sangsu Park 03 Jan '12

03 Jan '12

The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. --- sound/soc/soc-pcm.c | 38 ++++++++++++++++++-------------------- 1 files changed, 18 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..cdc860a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +641,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

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[alsa-devel] [PATCH] s3c24xx_simtec: fix typo
by Peter Meerwald 03 Jan '12

03 Jan '12

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> --- sound/soc/samsung/s3c24xx_simtec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index a253bcc..dcef50c 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -343,7 +343,7 @@ int __devinit simtec_audio_core_probe(struct platform_device *pdev, snd_dev = platform_device_alloc("soc-audio", -1); if (!snd_dev) { - dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + dev_err(&pdev->dev, "failed to alloc soc-audio device\n"); ret = -ENOMEM; goto err_gpio; } -- 1.7.4.1

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[alsa-devel] [PATCH] ALSA: HDA: Fix typo for ALC269VB_FIXUP_DMIC
by David Henningsson 03 Jan '12

03 Jan '12

This fixup is not actually used, so in practice this is just a cosmetic fix. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d24adbd..5e82acf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5320,7 +5320,7 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, - [ALC269_FIXUP_DMIC] = { + [ALC269VB_FIXUP_DMIC] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x12, 0x99a3092f }, /* int-mic */ -- 1.7.5.4

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[alsa-devel] [PATCH v3] ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs
by Sangsu Park 03 Jan '12

03 Jan '12

The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/soc-pcm.c | 38 ++++++++++++++++++-------------------- 1 files changed, 18 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..cdc860a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +641,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

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Re: [alsa-devel] It looks like snd-soc-rx51 only works as built-in, not as a module
by Mark Brown 02 Jan '12

02 Jan '12

On Mon, Jan 02, 2012 at 08:33:03PM +0200, Felipe Contreras wrote: > On Sat, Dec 31, 2011 at 2:59 AM, Mark Brown > > Which is relevant because...? To repeat, the ordering of module loading > > is totally ortogonal as any ordering of modules is supported. You > > appear to be totally ignoring what I am writing here. > One thing how things _should_ be, and another is how things actually > _are_. Right now snd-soc-rx51 doesn't work if it's loaded before > snd_soc_tlv320aic3x. Period. Can you provide logs showing this (with debug logging enabled in soc-core)? If this is happening it's a very serious issue, though I am rather surprised that nobody else has noticed such an issue since the usual instantiation order would have the card appear before the CODEC (due to the fact that the card is normally a platform device) and the structure of the code makes it difficult to imagine a failure path. > That means that *right now*, snd-soc-rx51 works fine with udev, > because snd_soc_tlv320aic3x will be loaded at boot time. Now, if > snd-soc-rx51 is converted and loaded automatically by udev as well, > the issues might appear again. Not if we look at and resolve the actual issue. Once again, any instantation ordering should be supported. I really don't understand why you keep on talking about fixing the module ordering,

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[alsa-devel] [PATCH v8 3/3 RESEND] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 02 Jan '12

02 Jan '12

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for AUX1, AUX2 and INPGA (Moved here from DAPM patch) Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 62e6a9c..ab38e93 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char * const da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] [PATCH 1/3] ASoC: Use dai_fmt in corgi machine driver
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/corgi.c | 14 ++------------ 1 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5e5004a..5ff6dac 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -142,18 +142,6 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, SND_SOC_CLOCK_IN); @@ -311,6 +299,8 @@ static struct snd_soc_dai_link corgi_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .init = corgi_wm8731_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &corgi_ops, }; -- 1.7.5.4

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