December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - One more Dell headset detection quirk
by Hui Wang 11 Dec '13

11 Dec '13

On the Dell machines with codec whose Subsystem Id is 0x10280624, no external microphone can be detected when plugging a 3-ring headset. If we add "model=dell-headset-multi" for the snd-hda-intel.ko, the problem will disappear. BugLink: https://bugs.launchpad.net/bugs/1259790 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- This patch was generated against the master branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 15300d3..c5f3e1b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5096,6 +5096,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), -- 1.8.1.2

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[alsa-devel] [PATCH] ALSA: hda - hdmi: Fix IEC958 ctl indexes for some simple HDMI devices
by Anssi Hannula 11 Dec '13

11 Dec '13

In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF outputs will have their IEC958 controls created starting from index 16 and the HDMI controls will be created starting from index 0. However, HDMI simple_playback_build_controls() as used by old VIA and NVIDIA codecs incorrectly requests the IEC958 controls to be created with an S/PDIF type instead of HDMI. In case the card has other codecs that have HDMI outputs, the controls will be created with wrong index=16, causing them to e.g. be unreachable by the ALSA "hdmi" alias. Fix that by making simple_playback_build_controls() request controls with HDMI indexes. Not many cards have an affected configuration, but e.g. ASUS M3N78-VM contains an integrated NVIDIA HDA "card" with: - a VIA codec that has, among others, an S/PDIF pin incorrectly labelled as an HDMI pin, and - an NVIDIA MCP7x HDMI codec. Reported-by: MysterX on #openelec Tested-by: MysterX on #openelec Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> # 3.8+ --- sound/pci/hda/patch_hdmi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 556d47231f11..f5060fc7c303 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2326,8 +2326,9 @@ static int simple_playback_build_controls(struct hda_codec *codec) int err; per_cvt = get_cvt(spec, 0); - err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid, - per_cvt->cvt_nid); + err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid, + per_cvt->cvt_nid, + HDA_PCM_TYPE_HDMI); if (err < 0) return err; return simple_hdmi_build_jack(codec, 0); -- 1.8.1.5

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[alsa-devel] fading in/out PCM audio
by Mat Gros 10 Dec '13

10 Dec '13

Hi, I'm using ALSA on Pandaboard. Is it possible to activate fading in/out ramps via the ALSA API? Greetings, Mat

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[alsa-devel] [PATCH] ALSA: at73c213: clk_round_rate() can return a zero upon error
by Paul Walmsley 10 Dec '13

10 Dec '13

Treat both negative and zero return values from clk_round_rate() as errors. This is needed since subsequent patches will convert clk_round_rate()'s return value to be an unsigned type, rather than a signed type, since some clock sources can generate rates higher than (2^31)-1 Hz. Eventually, when calling clk_round_rate(), only a return value of zero will be considered a error. All other values will be considered valid rates. The comparison against values less than 0 is kept to preserve the correct behavior in the meantime. Signed-off-by: Paul Walmsley <pwalmsley(a)nvidia.com> Cc: Hans-Christian Egtvedt <egtvedt(a)samfundet.no> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- Applies on v3.13-rc3. See also: http://marc.info/?l=linux-arm-kernel&m=138542591313620&w=2 sound/spi/at73c213.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 8e3d9a6c7a3b..25c38afaee49 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -174,7 +174,7 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) dac_rate_new = 8 * (ssc_rate / ssc_div); status = clk_round_rate(chip->board->dac_clk, dac_rate_new); - if (status < 0) + if (status <= 0) return status; /* Ignore difference smaller than 256 Hz. */

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[alsa-devel] [PATCH] ALSA: hda - Mute all aamix inputs as default
by Takashi Iwai 10 Dec '13

10 Dec '13

Not all channels have been initialized, so far, especially when aamix NID itself doesn't have amps but its leaves have. This patch fixes these holes. Otherwise you might get unexpected loopback inputs, e.g. from surround channels. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index c4671d00babd..249cd4615908 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4297,6 +4297,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return AC_PWRST_D3; } +/* mute all aamix inputs initially; parse up to the first leaves */ +static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) +{ + int i, nums; + const hda_nid_t *conn; + bool has_amp; + + nums = snd_hda_get_conn_list(codec, mix, &conn); + has_amp = nid_has_mute(codec, mix, HDA_INPUT); + for (i = 0; i < nums; i++) { + if (has_amp) + snd_hda_codec_amp_stereo(codec, mix, + HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); + else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) + snd_hda_codec_amp_stereo(codec, conn[i], + HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); + } +} /* * Parse the given BIOS configuration and set up the hda_gen_spec @@ -4435,6 +4455,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* mute all aamix input initially */ + if (spec->mixer_nid) + mute_all_mixer_nid(codec, spec->mixer_nid); + dig_only: parse_digital(codec); -- 1.8.5

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[alsa-devel] [PATCH V2 REPOST] ASoC: dmaengine: support deferred probe for DMA channels
by Stephen Warren 10 Dec '13

10 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Enhance dmaengine_pcm_request_chan_of() to support deferred probe for DMA channels, by using the new dma_request_slave_channel_or_err() API. This prevents snd_dmaengine_pcm_register() from succeeding without acquiring DMA channels due to the relevant DMA controller not yet being registered. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Mark, this is the patch which requires the dependency from the DMA tree, i.e. what I mentioned in the pull request that I sent just a little while ago. You'd previously ack'd it so I could take it through the Tegra tree, but it's probably better if instead you put it into ASoC tree first, then I pull your (topic/dma?) branch into the Tegra tree. (I've validated that your current topic/dma, plus a merge of the DMA tree's topic/defer_probe, plus this patch, plus a merge of next-20131210 to pick up the rest of ASoC etc., runs without conflicts and builds at least for Tegra without issues). v2: * Call dmaengine_pcm_release_chan() if dmaengine_pcm_request_chan_of() fails, to clean up any partial allocations. sound/soc/soc-generic-dmaengine-pcm.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5b70c556fba3..053c0fa9740d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -287,16 +287,17 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; -static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, +static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct device *dev, const struct snd_dmaengine_pcm_config *config) { unsigned int i; const char *name; + struct dma_chan *chan; if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || !dev->of_node) - return; + return 0; if (config->dma_dev) { /* @@ -318,13 +319,21 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = dmaengine_pcm_dma_channel_names[i]; if (config->chan_names[i]) name = config->chan_names[i]; - pcm->chan[i] = dma_request_slave_channel(dev, name); + chan = dma_request_slave_channel_reason(dev, name); + if (IS_ERR(chan)) { + if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER) + return -EPROBE_DEFER; + pcm->chan[i] = NULL; + } else + pcm->chan[i] = chan; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) break; } if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) pcm->chan[1] = pcm->chan[0]; + + return 0; } static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) @@ -360,7 +369,9 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev, config); + ret = dmaengine_pcm_request_chan_of(pcm, dev, config); + if (ret) + goto err_free_dma; if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) ret = snd_soc_add_platform(dev, &pcm->platform, -- 1.8.1.5

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[alsa-devel] [PATCH 0/5] ASoC: remove dmaengine compat_request_channel
by Stephen Warren 10 Dec '13

10 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> This series removes almost all users of the ASoC dmaengine compat_request_channel() function, and its implementation. The ux500 driver still uses compat_request_channel. I believe that Lars-Peter is working on removing its use from that driver. That patch or series would need to be inserted between 4/5 and 5/5 in this series. This series is based on ASoC's topic/dma branch. I have only compile-tested these patches. Stephen Warren (5): ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device ASoC: ep93xx: remove custom DMA alloc compat function ASoC: SPEAr: get rid of spear-pcm-audio struct device ASoC: SPEAr: remove custom DMA alloc compat function ASoC: dmaengine: remove compat_request_channel include/sound/dmaengine_pcm.h | 3 --- include/sound/spear_dma.h | 1 - sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 21 ++++++++++++++++-- sound/soc/cirrus/ep93xx-i2s.c | 23 +++++++++++++++++-- sound/soc/cirrus/ep93xx-pcm.c | 39 +++++++------------------------- sound/soc/cirrus/ep93xx-pcm.h | 24 ++++++++++++++++++++ sound/soc/cirrus/simone.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 3 --- sound/soc/spear/spdif_in.c | 25 +++++++++++++++++---- sound/soc/spear/spdif_out.c | 26 ++++++++++++++++++---- sound/soc/spear/spear_pcm.c | 42 ++++++++++------------------------- sound/soc/spear/spear_pcm.h | 26 ++++++++++++++++++++++ 14 files changed, 156 insertions(+), 83 deletions(-) create mode 100644 sound/soc/cirrus/ep93xx-pcm.h create mode 100644 sound/soc/spear/spear_pcm.h -- 1.8.1.5

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[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 10 Dec '13

10 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- This will need merging with the fix patch I sent earlier, I'l do that assuming this is OK. sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 7 +++++-- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 42 insertions(+), 45 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23a9893..4a88e36c82ec 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b494d4..ee23194f7ab8 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315a3eaf..fb09a1c5f75b 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..eab0050d4579 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { @@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256fc4b2c..6a5e4bf6ac96 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.5.1

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Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 10 Dec '13

10 Dec '13

> Codec: ATI R6xx HDMI >> Address: 0 >> AFG Function Id: 0x1 (unsol 0) >> Vendor Id: 0x1002aa01 >> Subsystem Id: 0x00aa0100 >> Revision Id: 0x100300 >> >> seem that your codec may get Audio-Video Association Port ID and Sink >> Information for your DENON-AVAMP >> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Revision ID 3 and later (0x100300 as reported by procfs codec#X) have support for providing more information than the previous revisions (but only if supported by the display driver). manufacture_id 0x0 product_id 0x0 port_id 0x0 you have to find out those values return by hda_codec_read ATI_VERB_GET_SINK_INFO_DATA as your codec are revision 3 and your graphic driver got the product_id, manufacture_id, port_id Manufacturer: DON Model 26 Serial Number 16843009 Made week 0 of 2011 Monitor name: DENON Audio data block Linear PCM, max channels 7 Supported sample rates (kHz): 192 176.4 96 88.2 48 44.1 32 Supported sample sizes (bits): 24 20 16 DTS, max channels 5 Supported sample rates (kHz): 96 88.2 48 44.1 Maximum bit rate: 1016 kHz AC-3, max channels 5 Supported sample rates (kHz): 48 44.1 32 Maximum bit rate: 1016 kHz DTS-HD, max channels 7 Supported sample rates (kHz): 192 176.4 96 88.2 48 44.1 Dolby Digital+, max channels 7 Supported sample rates (kHz): 48 44.1 MAT (MLP), max channels 7 Supported sample rates (kHz): 192 176.4 96 88.2 48 44.1 Speaker allocation data block Do those SAD in EDID in xorg.log match with the specification of your Denon HDMI receiver ? speakers [0x5f] FL/FR LFE FC RL/RR RC RLC/RRC sad_count 6 sad0_coding_type [0x1] LPCM sad0_channels 8 sad0_rates [0x1ee0] 32000 44100 48000 88200 96000 176400 192000 sad0_bits [0xe0000] 16 20 24 sad1_coding_type [0x2] AC-3 sad1_channels 6 sad1_rates [0xe0] 32000 44100 48000 sad1_max_bitrate 640000 sad2_coding_type [0x7] DTS sad2_channels 6 sad2_rates [0x6c0] 44100 48000 88200 96000 sad2_max_bitrate 1536000 sad3_coding_type [0xa] E-AC-3/DD+ (Dolby Digital Plus) sad3_channels 8 sad3_rates [0xc0] 44100 48000 sad4_coding_type [0xb] DTS-HD sad4_channels 8 sad4_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 sad5_coding_type [0xc] MLP (Dolby TrueHD) sad5_channels 8 sad5_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 seem emulated ELD 's max bitrate are different from those in SAD of EDID Do AC3 or DTS passthrough hdmi work (using alsa and pulseaudio) or not ? Do your Denon HDMI receiver support 7.1 ? Profiles: output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 400) output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3300) output:hdmi-stereo-extra1: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 200) output:hdmi-surround-extra1: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3100) output:hdmi-stereo-extra2: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 200) output:hdmi-surround-extra2: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3100) output:hdmi-stereo-extra3: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 200) output:hdmi-surround-extra3: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3100) seem no 7.1 in pulseaudio ? Do pulseaudio select the first HDMI as active port the default sink ? I am not sure. > seem no output of pactl stat http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/modules/alsa/mi… you may comment the last three hdmi devices until monitor_presence bug is fixed

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[alsa-devel] [PATCH] sound: memalloc.h - fix wrong truncation of dma_addr_t
by Stefano Panella 10 Dec '13

10 Dec '13

When running a 32bit kernel the hda_intel driver is still reporting a 64bit dma_mask if the HW supports it. >From sound/pci/hda/hda_intel.c: /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); else { pci_set_dma_mask(pci, DMA_BIT_MASK(32)); pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)); } which means when there is a call to dma_alloc_coherent from snd_malloc_dev_pages a machine address bigger than 32bit can be returned. This can be true in particular if running the 32bit kernel as a pv dom0 under the Xen Hypervisor or PAE on bare metal. The problem is that when calling setup_bdle to program the BLE the dma_addr_t returned from the dma_alloc_coherent is wrongly truncated from snd_sgbuf_get_addr if running a 32bit kernel: static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab, size_t offset) { struct snd_sg_buf *sgbuf = dmab->private_data; dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr; addr &= PAGE_MASK; return addr + offset % PAGE_SIZE; } where PAGE_MASK in a 32bit kernel is zeroing the upper 32bit af addr. Without this patch the HW will fetch the 32bit truncated address, which is not the one obtained from dma_alloc_coherent and will result to a non working audio but can corrupt host memory at a random location. The current patch apply to v3.13-rc3-74-g6c843f5 Signed-off-by: Stefano Panella <stefano.panella(a)citrix.com> Reviewed-by: Frediano Ziglio <frediano.ziglio(a)citrix.com> --- include/sound/memalloc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index af99839..5f73785 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -108,7 +108,7 @@ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab, { struct snd_sg_buf *sgbuf = dmab->private_data; dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr; - addr &= PAGE_MASK; + addr &= ~((dma_addr_t)PAGE_SIZE - 1); return addr + offset % PAGE_SIZE; } -- 1.7.9.5

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