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

[alsa-devel] [PATCH] hdspm.c: make driver name generic to avoid user confusion
by Jörn Nettingsmeier 07 Jun '11

07 Jun '11

hi adi, *! here's a trivial patch to change the driver name of hdspm.c to avoid irritating users who don't have the MADI version (just happened on the LAU list). iiuc, no external software should make any assumptions about this string, so there should be no side effects, but i have been wrong before :) please review and apply. best, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

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[alsa-devel] [PATCH] ALSA: hda: Fix quirk for Dell Inspiron 910
by Daniel Chen 07 Jun '11

07 Jun '11

Hi, Attached is a patch against sound-2.6 git HEAD that resolves inaudible audio from internal speakers on the reporter's Dell Inspiron 910. Please apply. Cheers, -Dan

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[alsa-devel] alsa-driver build problem : no lola dir
by Eliot Blennerhassett 07 Jun '11

07 Jun '11

(resend from address subscribed to alsa-devel) Trying to build alsa-driver git fails: make[2]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/pci/korg1212' make: Entering an unknown directory make: *** lola: No such file or directory. Stop. make: Leaving an unknown directory make[1]: *** [prepare] Error 1 make[1]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/pci' make: *** [dep] Error 1 ~/src/alsagit/alsa-driver$ my alsa-kernel links point to a current sound-2.6 git tree. -- Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] [PATCH 1/6] ASoC: AD1836: Fix setting the PCM format
by Lars-Peter Clausen 06 Jun '11

06 Jun '11

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This patch doesn't need to go into stable since there are currently no users of PCM formats other than S32, which is the default. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 14 +++++++------- sound/soc/codecs/ad1836.h | 6 ++++++ 2 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ab63d52..754c496 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; + word_len = AD1836_WORD_LEN_16; break; case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; + word_len = AD1836_WORD_LEN_20; break; case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; + word_len = AD1836_WORD_LEN_24; break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, - AD1836_DAC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, - AD1836_ADC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + word_len << AD1836_ADC_WORD_OFFSET); return 0; } diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 8455967..9d6a3f8 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -25,6 +25,7 @@ #define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) #define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) #define AD1836_DAC_WORD_LEN_MASK 0x18 +#define AD1836_DAC_WORD_LEN_OFFSET 3 #define AD1836_DAC_CTRL2 1 #define AD1836_DACL1_MUTE 0 @@ -51,6 +52,7 @@ #define AD1836_ADCL2_MUTE 2 #define AD1836_ADCR2_MUTE 3 #define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_WORD_OFFSET 5 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) @@ -60,4 +62,8 @@ #define AD1836_NUM_REGS 16 +#define AD1836_WORD_LEN_24 0x0 +#define AD1836_WORD_LEN_20 0x1 +#define AD1836_WORD_LEN_16 0x2 + #endif -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: Allow WM8915 BCLK calculation outside hw_params()
by Mark Brown 06 Jun '11

06 Jun '11

Allow more dynamic management of the device clocking by allowing BCLK to be calculated when we set SYSCLK. This means that if the system is idle when hw_params() runs then we don't try to use the SYSCLK used in that case to set up the BCLK dividers, we can instead wait until a later point such as bias level configuration. This makes it easier to manage low power modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 82 +++++++++++++++++++++++++++++---------------- 1 files changed, 53 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index a1d8618..423baa9 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -75,6 +75,7 @@ struct wm8915_priv { struct wm8915_pdata pdata; int rx_rate[WM8915_AIFS]; + int bclk_rate[WM8915_AIFS]; /* Platform dependant ReTune mobile configuration */ int num_retune_mobile_texts; @@ -1562,6 +1563,50 @@ static int wm8915_reset(struct snd_soc_codec *codec) return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915); } +static const int bclk_divs[] = { + 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 +}; + +static void wm8915_update_bclk(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int aif, best, cur_val, bclk_rate, bclk_reg, i; + + /* Don't bother if we're in a low frequency idle mode that + * can't support audio. + */ + if (wm8915->sysclk < 64000) + return; + + for (aif = 0; aif < WM8915_AIFS; aif++) { + switch (aif) { + case 0: + bclk_reg = WM8915_AIF1_BCLK; + break; + case 1: + bclk_reg = WM8915_AIF2_BCLK; + break; + } + + bclk_rate = wm8915->bclk_rate[aif]; + + /* Pick a divisor for BCLK as close as we can get to ideal */ + best = 0; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate; + if (cur_val < 0) /* BCLK table is sorted */ + break; + best = i; + } + bclk_rate = wm8915->sysclk / bclk_divs[best]; + dev_dbg(codec->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", + bclk_divs[best], bclk_rate); + + snd_soc_update_bits(codec, bclk_reg, + WM8915_AIF1_BCLK_DIV_MASK, best); + } +} + static int wm8915_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1714,10 +1759,6 @@ static int wm8915_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static const int bclk_divs[] = { - 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 -}; - static const int dsp_divs[] = { 48000, 32000, 16000, 8000 }; @@ -1728,17 +1769,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); - int bits, i, bclk_rate, best, cur_val; + int bits, i, bclk_rate; int aifdata = 0; - int bclk = 0; int lrclk = 0; int dsp = 0; - int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift; - - if (!wm8915->sysclk) { - dev_err(codec->dev, "SYSCLK not configured\n"); - return -EINVAL; - } + int aifdata_reg, lrclk_reg, dsp_shift; switch (dai->id) { case 0: @@ -1750,7 +1785,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1; lrclk_reg = WM8915_AIF1_TX_LRCLK_1; } - bclk_reg = WM8915_AIF1_BCLK; dsp_shift = 0; break; case 1: @@ -1762,7 +1796,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1; lrclk_reg = WM8915_AIF2_TX_LRCLK_1; } - bclk_reg = WM8915_AIF2_BCLK; dsp_shift = WM8915_DSP2_DIV_SHIFT; break; default: @@ -1776,6 +1809,9 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, return bclk_rate; } + wm8915->bclk_rate[dai->id] = bclk_rate; + wm8915->rx_rate[dai->id] = params_rate(params); + /* Needs looking at for TDM */ bits = snd_pcm_format_width(params_format(params)); if (bits < 0) @@ -1793,18 +1829,7 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, } dsp |= i << dsp_shift; - /* Pick a divisor for BCLK as close as we can get to ideal */ - best = 0; - for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { - cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate; - if (cur_val < 0) /* BCLK table is sorted */ - break; - best = i; - } - bclk_rate = wm8915->sysclk / bclk_divs[best]; - dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", - bclk_divs[best], bclk_rate); - bclk |= best; + wm8915_update_bclk(codec); lrclk = bclk_rate / params_rate(params); dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", @@ -1814,14 +1839,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, WM8915_AIF1TX_WL_MASK | WM8915_AIF1TX_SLOT_LEN_MASK, aifdata); - snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk); snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK, lrclk); snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2, WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp); - wm8915->rx_rate[dai->id] = params_rate(params); - return 0; } @@ -1882,6 +1904,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, return -EINVAL; } + wm8915_update_bclk(codec); + snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK, src << WM8915_SYSCLK_SRC_SHIFT | ratediv); -- 1.7.5.3

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[alsa-devel] [PATCH 1/6] ASoC: Indentation fix for null loop operation
by Mark Brown 06 Jun '11

06 Jun '11

More with the legibility. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6be6546..988cdff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -209,7 +209,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val, item, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; + ; val = snd_soc_read(w->codec, e->reg); item = (val >> e->shift_l) & (bitmask - 1); -- 1.7.5.3

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[alsa-devel] [PATCH 1/2] ASoC: Specify target bias state directly as a bias state
by Mark Brown 06 Jun '11

06 Jun '11

Rather than a simple flag to say if we want the DAPM context to be at full power specify the target bias state. This should have no current effect but is a bit more direct and so makes it easier to change our decisions about the which bias state to go into in future. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 8 +++--- sound/soc/soc-dapm.c | 60 +++++++++++++++++++++++++-------------------- 3 files changed, 38 insertions(+), 32 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c46e7d8..7415878 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,7 +510,7 @@ struct snd_soc_dapm_context { struct snd_soc_card *card; /* parent card */ /* used during DAPM updates */ - int dev_power; + enum snd_soc_bias_level target_bias_level; struct list_head list; #ifdef CONFIG_DEBUG_FS diff --git a/include/sound/soc.h b/include/sound/soc.h index f1de3e0..0f29700 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -214,10 +214,10 @@ * @OFF: Power Off. No restrictions on transition times. */ enum snd_soc_bias_level { - SND_SOC_BIAS_OFF, - SND_SOC_BIAS_STANDBY, - SND_SOC_BIAS_PREPARE, - SND_SOC_BIAS_ON, + SND_SOC_BIAS_OFF = 0, + SND_SOC_BIAS_STANDBY = 1, + SND_SOC_BIAS_PREPARE = 2, + SND_SOC_BIAS_ON = 3, }; struct snd_jack; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 776e6f4..0bbded4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1042,16 +1042,17 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) struct snd_soc_dapm_context *d = data; int ret; - if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) { + /* If we're off and we're not supposed to be go into STANDBY */ + if (d->bias_level == SND_SOC_BIAS_OFF && + d->target_bias_level != SND_SOC_BIAS_OFF) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, "Failed to turn on bias: %d\n", ret); } - /* If we're changing to all on or all off then prepare */ - if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) || - (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) { + /* Prepare for a STADDBY->ON or ON->STANDBY transition */ + if (d->bias_level != d->target_bias_level) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, @@ -1068,7 +1069,9 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) int ret; /* If we just powered the last thing off drop to standby bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) { + if (d->bias_level == SND_SOC_BIAS_PREPARE && + (d->target_bias_level == SND_SOC_BIAS_STANDBY || + d->target_bias_level == SND_SOC_BIAS_OFF)) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, "Failed to apply standby bias: %d\n", @@ -1076,14 +1079,16 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } /* If we're in standby and can support bias off then do that */ - if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) { + if (d->bias_level == SND_SOC_BIAS_STANDBY && + d->target_bias_level == SND_SOC_BIAS_OFF) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); if (ret != 0) dev_err(d->dev, "Failed to turn off bias: %d\n", ret); } /* If we just powered up then move to active bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) { + if (d->bias_level == SND_SOC_BIAS_PREPARE && + d->target_bias_level == SND_SOC_BIAS_ON) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON); if (ret != 0) dev_err(d->dev, "Failed to apply active bias: %d\n", @@ -1108,13 +1113,19 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(up_list); LIST_HEAD(down_list); LIST_HEAD(async_domain); + enum snd_soc_bias_level bias; int power; trace_snd_soc_dapm_start(card); - list_for_each_entry(d, &card->dapm_list, list) - if (d->n_widgets || d->codec == NULL) - d->dev_power = 0; + list_for_each_entry(d, &card->dapm_list, list) { + if (d->n_widgets || d->codec == NULL) { + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; + } + } /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. @@ -1137,7 +1148,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) else power = 1; if (power) - w->dapm->dev_power = 1; + w->dapm->target_bias_level = SND_SOC_BIAS_ON; if (w->power == power) continue; @@ -1161,24 +1172,19 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) switch (event) { case SND_SOC_DAPM_STREAM_START: case SND_SOC_DAPM_STREAM_RESUME: - dapm->dev_power = 1; + dapm->target_bias_level = SND_SOC_BIAS_ON; break; case SND_SOC_DAPM_STREAM_STOP: - dapm->dev_power = !!dapm->codec->active; + if (dapm->codec->active) + dapm->target_bias_level = SND_SOC_BIAS_ON; + else + dapm->target_bias_level = SND_SOC_BIAS_STANDBY; break; case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->dev_power = 0; + dapm->target_bias_level = SND_SOC_BIAS_STANDBY; break; case SND_SOC_DAPM_STREAM_NOP: - switch (dapm->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - dapm->dev_power = 0; - break; - default: - dapm->dev_power = 1; - break; - } + dapm->target_bias_level = dapm->bias_level; break; default: break; @@ -1186,12 +1192,12 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } /* Force all contexts in the card to the same bias state */ - power = 0; + bias = SND_SOC_BIAS_OFF; list_for_each_entry(d, &card->dapm_list, list) - if (d->dev_power) - power = 1; + if (d->target_bias_level > bias) + bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - d->dev_power = power; + d->target_bias_level = bias; /* Run all the bias changes in parallel */ -- 1.7.5.3

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Re: [alsa-devel] [PATCH] snd-usb-audio with multiple audio functions
by Clemens Ladisch 06 Jun '11

06 Jun '11

Rolf Meeser wrote: > On 06/06/2011 01:44 PM, Clemens Ladisch wrote: > > Many audio devices with multiple vendor-specific interfaces do not have > > a control interface. This code would split them into multiple cards. > > Breaking existing setups would indeed be bad. > Are that cards with audio interfaces not following the Audio Class spec? Yes. > What would be the easiest way to identify them, so we could assign a new > card index only in case of a compliant card? Non-compliant devices don't have a control interface. In theory, the USB core probes a device's interfaces in order, so the control interface usually is probed first. However, I'm not sure if this is really true for all devices. It's probably safest to check explicitly if the device has any interface marked as audio control 1/2. Regards, Clemens

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[alsa-devel] [PATCH] snd-usb-audio with multiple audio functions
by Rolf Meeser 06 Jun '11

06 Jun '11

Hello, This patch fixes the currently broken support for multiple audio functions (AF) in snd-usb-audio. The patch is against the stable 2.6.39.1 kernel. It addresses three issues: 1. Controls of the second and any further AF are currently not accessibly (controls of the first AF are duplicated!) 2. Multiple AF's are currently incorrectly combined into one sound card index. 3. Module usage count of snd-usb-audio does not drop to zero after disconnecting a multiple AF device. All three issues are fixed by assigning a new card index for each AF in a USB device. Rationale: - A USB audio function (AF) consists of a control interface and zero or more streaming interfaces. USB devices can have multiple AF, where the individual AF are completely independent of each other. In particular, it cannot be assumed that the controls of one AF can be manipulated through the control interface of another (the first) AF. - Older kernels (up to 2.6.32? at least before the major rewrite of USB audio code) supported multiple AF's. However, they combined the streams into one sound card: For instance hw:1,0 for the stream interface of the first AF, and hw:1,1 for the stream interface of the second AF. I believe this is incorrect, as different AF's must be treated as an independent entity. Each AF must get its own card index, for the example this would be hw:1,0 and hw:2,0. This is also the way THE OTHER OS implements it. - Newer kernels have broken multiple AF support. In a dual AF setup these kernels detect the second AF, but they duplicate the controls of the first AF, and ignore the controls of the second one! - The patch also fixes a problem with module usage: When you unplug a multiple AF device, the snd-usb-audio module usage count currently stays at 1, while you could expect it to go to zero. This is the first time I'm sending a patch to this list. Please be patient with me if I did something in a wrong way :-) Regards, Rolf

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[alsa-devel] Fwd: Re: [PATCH] snd-usb-audio with multiple audio functions
by Rolf Meeser 06 Jun '11

06 Jun '11

Hi Clemens, On 06/06/2011 01:44 PM, Clemens Ladisch wrote: > Rolf Meeser wrote: >> + if (usb_chip[i]&& >> + (usb_chip[i]->dev == dev)&& >> + (usb_chip[i]->ctrl_intf)&& >> + (usb_chip[i]->ctrl_intf->desc.bInterfaceNumber == ifnum)) { > Many audio devices with multiple vendor-specific interfaces do not have > a control interface. This code would split them into multiple cards. > > > Regards, > Clemens > Breaking existing setups would indeed be bad! Can you see an easy way to identify such cards here? Then we could assign a new card index only in case of a class compliant card, where this is desirable. Regards, Rolf

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