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

[alsa-devel] [PATCH] ASoC: Support WM8994 mono AIF configurations
by Mark Brown 08 Dec '10

08 Dec '10

The WM8994 supports mono signals - enable this in the driver. With DSP mode an automatic data channel selector is available, activate this when in mono mode. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 21 +++++++++++++++------ 1 files changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 78b96f6..da48802 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2121,10 +2121,12 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; + int aif2_reg; int bclk_reg; int lrclk_reg; int rate_reg; int aif1 = 0; + int aif2 = 0; int bclk = 0; int lrclk = 0; int rate_val = 0; @@ -2135,6 +2137,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, switch (dai->id) { case 1: aif1_reg = WM8994_AIF1_CONTROL_1; + aif2_reg = WM8994_AIF1_CONTROL_2; bclk_reg = WM8994_AIF1_BCLK; rate_reg = WM8994_AIF1_RATE; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || @@ -2147,6 +2150,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, break; case 2: aif1_reg = WM8994_AIF2_CONTROL_1; + aif2_reg = WM8994_AIF2_CONTROL_2; bclk_reg = WM8994_AIF2_BCLK; rate_reg = WM8994_AIF2_RATE; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || @@ -2202,6 +2206,10 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n", dai->id, wm8994->aifclk[id], bclk_rate); + if (params_channels(params) == 1 && + (snd_soc_read(codec, aif1_reg) & 0x18) == 0x18) + aif2 |= WM8994_AIF1_MONO; + if (wm8994->aifclk[id] == 0) { dev_err(dai->dev, "AIF%dCLK not configured\n", dai->id); return -EINVAL; @@ -2245,6 +2253,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, lrclk, bclk_rate / lrclk); snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); + snd_soc_update_bits(codec, aif2_reg, WM8994_AIF1_MONO, aif2); snd_soc_update_bits(codec, bclk_reg, WM8994_AIF1_BCLK_DIV_MASK, bclk); snd_soc_update_bits(codec, lrclk_reg, WM8994_AIF1DAC_RATE_MASK, lrclk); @@ -2400,14 +2409,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .id = 1, .playback = { .stream_name = "AIF1 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, .capture = { .stream_name = "AIF1 Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, @@ -2419,14 +2428,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .id = 2, .playback = { .stream_name = "AIF2 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, .capture = { .stream_name = "AIF2 Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, @@ -2438,14 +2447,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .id = 3, .playback = { .stream_name = "AIF3 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, .capture = { .stream_name = "AIF3 Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: Correct WM8962 interrupt mask register read
by Mark Brown 08 Dec '10

08 Dec '10

Fix mismerge from the out of tree BSP where this support was developed. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e809274..1304ca9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3339,7 +3339,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) int mask; int active; - mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); + mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; -- 1.7.1

2 1

[PATCH] Fix possible sprintf overrun in snd_pcm_hw_open
by David Henningsson 08 Dec '10

08 Dec '10

BugLink: http://launchpad.net/bugs/668487 Possible buffer overrun if the number of "card" and "device" are absurdly high, especially on 64-bit platforms. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_hw.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 9d243d5..ce74ad4 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1270,7 +1270,7 @@ int snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, SNDERR("invalid stream %d", stream); return -EINVAL; } - sprintf(filename, filefmt, card, device); + snprintf(filename, sizeof(filename), filefmt, card, device); __again: if (attempt++ > 3) { -- 1.7.1 --------------080408060601020802060106--

1 0

[alsa-devel] [PATCH 1/3] ASoC: SAMSUNG: Fix initial return value
by Seungwhan Youn 08 Dec '10

08 Dec '10

This patch fixed intial return value to be a '0' as asuccess on set_audio_clock_heirachy(). This avoids unintended error on initialize. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/smdk_spdif.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c index c8bd904..cb658d1 100644 --- a/sound/soc/s3c24xx/smdk_spdif.c +++ b/sound/soc/s3c24xx/smdk_spdif.c @@ -28,7 +28,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev) { struct clk *fout_epll, *mout_epll, *sclk_audio0, *sclk_spdif; - int ret; + int ret = 0; fout_epll = clk_get(NULL, "fout_epll"); if (IS_ERR(fout_epll)) { -- 1.6.2.5

4 19

[alsa-devel] [PATCH] ASoC: WM8580: Debug BCLK and sample size
by Jassi Brar 08 Dec '10

08 Dec '10

In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32, rather than WM8580_AIF_LENGTH_24. Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/codecs/wm8580.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index cb7765f..45efc94 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -491,16 +491,16 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, paifa |= 0x8; break; case SNDRV_PCM_FORMAT_S20_3LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_20; break; case SNDRV_PCM_FORMAT_S24_LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_24; break; case SNDRV_PCM_FORMAT_S32_LE: - paifa |= 0x10; - paifb |= WM8580_AIF_LENGTH_24; + paifa |= 0x0; + paifb |= WM8580_AIF_LENGTH_32; break; default: return -EINVAL; -- 1.6.2.5

4 4

[alsa-devel] [PATCH] ALSA: hda - Reset sample sizes and max bitrates when reading ELD
by Anssi Hannula 08 Dec '10

08 Dec '10

When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc() is called for every SAD (Short Audio Descriptor) in the ELD data. For LPCM coding type SAD defines the supported sample sizes. For several other coding types (such as AC-3), a maximum bitrate is defined. The maximum bitrate and sample size fields are not always cleared. Therefore, if a device is unplugged and a different one is plugged in, and the coding types of some SAD positions differ between the devices, the old max_bitrate or sample_bits values will persist if the new SADs do not define those values. The leftover max_bitrate and sample_bits do not cause any issues other than wrongly showing up in eld#X.Y procfs file and kernel log. Fix that by always clearing sample_bits and max_bitrate when reading SADs. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/hda_eld.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 009031f..4a66347 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -189,6 +189,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->channels = GRAB_BITS(buf, 0, 0, 3); a->channels++; + a->sample_bits = 0; + a->max_bitrate = 0; + a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: @@ -198,7 +201,6 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, case AUDIO_CODING_TYPE_LPCM: val = GRAB_BITS(buf, 2, 0, 3); - a->sample_bits = 0; for (i = 0; i < 3; i++) if (val & (1 << i)) a->sample_bits |= cea_sample_sizes[i + 1]; -- 1.7.3

2 2

Re: [alsa-devel] [PATCH] ALSA: hda - Do not wrongly restrict min_channels based on ELD
by Stephen Warren 08 Dec '10

08 Dec '10

Anssi Hannula wrote: > Commit bbbe33900d1f3c added functionality to restrict PCM parameters > based on ELD info (derived from EDID data) of the audio sink. > > However, it wrongly assumes that the bits 0-2 of the first byte of > CEA Short Audio Descriptors mean a supported number of channels. In > reality, they mean the maximum number of channels (as per CEA-861-D > 7.5.2). This means that the channel count can only be used to restrict > max_channels, not min_channels. > > Restricting min_channels causes us to deny opening the device in stereo > mode if the sink only has SADs that declare larger numbers of channels > (like Primare SP32 AV Processor does). > > Fix that by not restricting min_channels based on ELD information. Yes, re-reading the CEA-861-D spec, that makes sense. Comments inline though. BTW, I noticed that no SADs are required if only basic audio is supported. Is that case handled by the ELD parser? I.e. instead of: pcm->channels_max = 0; Should it be: pcm->channels_max = 0; if supports basic audio: pcm->channels_max = 2; or something like that? > Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> > Reported-by: Jean-Yves Avenard <jyavenard(a)gmail.com> > Tested-by: Jean-Yves Avenard <jyavenard(a)gmail.com> > Cc: stable(a)kernel.org > --- > > I think this should go to 2.6.36 stable tree as well. It does not affect > earlier stable releases. > > sound/pci/hda/hda_eld.c | 4 ---- > sound/pci/hda/patch_hdmi.c | 1 - > 2 files changed, 0 insertions(+), 5 deletions(-) > > diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c > index cb0c23a..47ef8aa 100644 > --- a/sound/pci/hda/hda_eld.c > +++ b/sound/pci/hda/hda_eld.c > @@ -601,13 +601,10 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct > hda_pcm_stream *pcm, > pcm->rates = 0; > pcm->formats = 0; > pcm->maxbps = 0; > - pcm->channels_min = -1; > pcm->channels_max = 0; > for (i = 0; i < eld->sad_count; i++) { > struct cea_sad *a = &eld->sad[i]; > pcm->rates |= a->rates; > - if (a->channels < pcm->channels_min) > - pcm->channels_min = a->channels; > if (a->channels > pcm->channels_max) > pcm->channels_max = a->channels; > if (a->format == AUDIO_CODING_TYPE_LPCM) { That all looks OK. > @@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct > hda_pcm_stream *pcm, > /* restrict the parameters by the values the codec provides */ > pcm->rates &= codec_pars->rates; > pcm->formats &= codec_pars->formats; > - pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); Shouldn't that be: pcm->channels_min = pcm->channels_min; Or, is that assignment already made earlier as a default? > pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); > pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); > } > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > index d3e49aa..31df774 100644 > --- a/sound/pci/hda/patch_hdmi.c > +++ b/sound/pci/hda/patch_hdmi.c > @@ -834,7 +834,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, > return -ENODEV; > } else { > /* fallback to the codec default */ > - hinfo->channels_min = codec_pars->channels_min; Isn't this still required to default the value? > hinfo->channels_max = codec_pars->channels_max; > hinfo->rates = codec_pars->rates; > hinfo->formats = codec_pars->formats; > -- > 1.7.3 -- nvpublic

2 5

[alsa-devel] [PATCH] ALSA: hda - Always allow basic audio irrespective of ELD info
by Anssi Hannula 07 Dec '10

07 Dec '10

Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, according to CEA-861-D no SAD is needed for basic audio (32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a basic audio flag in the CEA EDID Extension. The flag is not present in ELD. However, as all audio capable sinks are required to support basic audio, we can assume it to be always available. Fix allowed audio formats with sinks that have SADs (Short Audio Descriptors) which do not completely overlap with the basic audio formats (there are no reports of affected devices so far) by always assuming that basic audio is supported. Reported-by: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- Tested on 2.6.36 and I saw no obvious regressions. The issue affects stable 2.6.36 as well, but as there are no reports of affected devices, I'll leave the decision on whether to send this to stable@ for the ALSA subsystem maintainers. sound/pci/hda/hda_eld.c | 16 +++++++--------- 1 files changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 47ef8aa..009031f 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -598,21 +598,19 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, { int i; - pcm->rates = 0; - pcm->formats = 0; - pcm->maxbps = 0; - pcm->channels_max = 0; + /* assume basic audio support (the basic audio flag is not in ELD; + * however, all audio capable sinks are required to support basic + * audio) */ + pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; + pcm->formats = SNDRV_PCM_FMTBIT_S16_LE; + pcm->maxbps = 16; + pcm->channels_max = 2; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { - if (a->sample_bits & AC_SUPPCM_BITS_16) { - pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE; - if (pcm->maxbps < 16) - pcm->maxbps = 16; - } if (a->sample_bits & AC_SUPPCM_BITS_20) { pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; if (pcm->maxbps < 20) -- 1.7.3

2 3

[alsa-devel] [PATCH] ALSA: hda - Do not wrongly restrict min_channels based on ELD
by Anssi Hannula 07 Dec '10

07 Dec '10

Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, it wrongly assumes that the bits 0-2 of the first byte of CEA Short Audio Descriptors mean a supported number of channels. In reality, they mean the maximum number of channels (as per CEA-861-D 7.5.2). This means that the channel count can only be used to restrict max_channels, not min_channels. Restricting min_channels causes us to deny opening the device in stereo mode if the sink only has SADs that declare larger numbers of channels (like Primare SP32 AV Processor does). Fix that by not restricting min_channels based on ELD information. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Reported-by: Jean-Yves Avenard <jyavenard(a)gmail.com> Tested-by: Jean-Yves Avenard <jyavenard(a)gmail.com> Cc: stable(a)kernel.org --- I think this should go to 2.6.36 stable tree as well. It does not affect earlier stable releases. sound/pci/hda/hda_eld.c | 4 ---- sound/pci/hda/patch_hdmi.c | 1 - 2 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index cb0c23a..47ef8aa 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -601,13 +601,10 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->rates = 0; pcm->formats = 0; pcm->maxbps = 0; - pcm->channels_min = -1; pcm->channels_max = 0; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; - if (a->channels < pcm->channels_min) - pcm->channels_min = a->channels; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { @@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, /* restrict the parameters by the values the codec provides */ pcm->rates &= codec_pars->rates; pcm->formats &= codec_pars->formats; - pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d3e49aa..31df774 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -834,7 +834,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, return -ENODEV; } else { /* fallback to the codec default */ - hinfo->channels_min = codec_pars->channels_min; hinfo->channels_max = codec_pars->channels_max; hinfo->rates = codec_pars->rates; hinfo->formats = codec_pars->formats; -- 1.7.3

1 0

[alsa-devel] [PATCH] ASoC: soc-core: Fix null pointer dereference
by Dimitris Papastamos 07 Dec '10

07 Dec '10

In case the codec driver did not provide a read/write function, codec->driver->read|write will be NULL. Ensure that we use the one specified in codec->read|write to avoid oopsing when we access the debugfs entries. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 392d336..078f9df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -117,7 +117,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) * the register being volatile and the device being * powered off. */ - ret = codec->driver->read(codec, i); + ret = codec->read(codec, i); if (ret >= 0) count += snprintf(buf + count, PAGE_SIZE - count, @@ -228,7 +228,7 @@ static ssize_t codec_reg_write_file(struct file *file, start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; - codec->driver->write(codec, reg, value); + codec->write(codec, reg, value); return buf_size; } -- 1.7.3.3

2 1