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

Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 07 Feb '13

07 Feb '13

On Thu, Feb 07, 2013 at 01:15:54AM +0000, Richard Fitzgerald wrote: > > +- partial drain > > +This is called when end of file is reached. The userspace can inform DSP that > > +EOF is reached and now DSP can start skipping padding delay. Also next write > > +data would belong to next track > > We're really doing two different tasks inside the "partial drain". The name, > and the reason we are doing two tasks in one, comes from a particular higher- > layer scenario, but there's no reason the driver API need use the same > terminology as one particular application of the functionality. > The two tasks are: > > 1) Tell the DSP that we have sent all data for the current track and following > data will be for the next track. This lets the DSP lay down a marker for where > it should strip padding at the end of a track, and know it should be expecting > more data for another track to follow gaplessly where it must strip the > encoder delay from the start. > > 2) Ask for notification when DSP has reached the changeover point between the > playback of the two tracks > > I think it would be more logical and less confusing not to combine the two into > a single ioctl. Instead, add only one new ioctl specifically to provide the > track changeover hint, something like SNDRV_COMPRESS_NEXT_TRACK (meaning DSP > should expect data for next track to follow). Don't add a new drain, just use > the existing SNDRV_COMPRESS_DRAIN - the driver/DSP can make a decision whether > it needs to do something special with the drain if we have told it that we > will be sending it some more data for a following track. > > So the SNDRV_COMPRESS_PARTIAL_DRAIN in this patch would become > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > 2) Send SNDRV_COMPRESS_DRAIN The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which expects the decoder to completely drain its buffers and come to complete halt. This would also mean the framework will treat a drained stream as stopped and needs a new start. Certainly we dont want that in this case. So we can't use SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that would overtly complicate this. If we are not doing proper drian lets not call it that. But I think I like the idea of splitting the two up to do a cleaner interface. Let me check this... > When we reach the final track we just do SNDRV_COMPRESS_DRAIN to wait for it > to finish. > > If we setup a next track by doing SNDRV_COMPRESS_NEXT_TRACK and then change > our mind and decide that this is going to be the last track, we do a > SNDRV_COMPRESS_DRAIN without sending any next track data, then we do > SNDRV_COMPRESS_STOP Nope that is where we would have issue. You dont call SNDRV_COMPRESS_STOP for compressed streams. You have to wait till DSP has decoded and rendered data for the last track which can be done by using SNDRV_COMPRESS_DRAIN only. SNDRV_COMPRESS_STOP should not be called in this case. -- ~Vinod

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[alsa-devel] [PATCH] hda-analyzer: Make parsing code more robust against future proc file changes
by David Henningsson 07 Feb '13

07 Feb '13

Historically, new additions to the codec proc file has crashed hda-analyzer. With this patch, new additions will instead be ignored. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-analyzer/hda_proc.py | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/hda-analyzer/hda_proc.py b/hda-analyzer/hda_proc.py index 3d544c8..10aa662 100644 --- a/hda-analyzer/hda_proc.py +++ b/hda-analyzer/hda_proc.py @@ -644,9 +644,8 @@ class HDACodecProc(HDACodec, HDABaseProc): if line == 'Invalid AFG subtree': print "Invalid AFG subtree for codec %s?" % self.proc_codec_id return - while line.startswith('Power-Map: ') or \ - line.startswith('Analog Loopback: '): - print 'Sigmatel specific "%s" verb ignored for the moment' % line + while not line.startswith('Node '): + print 'Vendor specific "%s" verb ignored at the moment' % line idx += 1 line = lines[idx].strip() node = None @@ -728,7 +727,7 @@ class HDACodecProc(HDACodec, HDABaseProc): elif line.startswith(' Volume-Knob: '): node.add_volknob(line[15:]) else: - self.wrongfile(line) + print 'Node line "%s" ignored at the moment' % line idx += 1 if idx < len(lines): line = lines[idx] -- 1.7.9.5

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[alsa-devel] [PATCH] hda-emu: Add sanify check of initial pinctl
by David Henningsson 07 Feb '13

07 Feb '13

The pinctl in alsa-info is not the initial pinctl but the current, so sanify the current pinctl before accepting it as the initial pinctl. (The initial pinctl is used during pm test.) Several false positives in the test tool is removed with this improvement. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-int.c | 64 +++++++++++++++++++++++++++++++++++---------------- hda-parse.c | 2 ++ include/hda-types.h | 4 ++++ 3 files changed, 50 insertions(+), 20 deletions(-) diff --git a/hda-int.c b/hda-int.c index c25af56..fbd3cee 100644 --- a/hda-int.c +++ b/hda-int.c @@ -400,29 +400,37 @@ static int get_channel_streamid(struct xhda_codec *codec, return node->streamid; } -static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, - unsigned int cmd) +void hda_verify_pin_ctl(struct xhda_node *node, int log, + unsigned int *sanified_pinctl) { - if (!node) - return 0; - node->pinctl = cmd & 0xff; + int pinctl = node->pinctl; /* sanity checks */ if ((node->pinctl & AC_PINCTL_OUT_EN) && - !(node->pincap & AC_PINCAP_OUT)) - hda_log(HDA_LOG_ERR, - "setting OUT_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_OUT)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting OUT_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_OUT_EN; + } if ((node->pinctl & AC_PINCTL_HP_EN) && - !(node->pincap & AC_PINCAP_HP_DRV)) - hda_log(HDA_LOG_ERR, - "setting HP_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_HP_DRV)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting HP_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_HP_EN; + } if ((node->pinctl & AC_PINCTL_IN_EN) && - !(node->pincap & AC_PINCAP_IN)) - hda_log(HDA_LOG_ERR, - "setting IN_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_IN)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting IN_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_IN_EN; + } + if (node->pinctl & AC_PINCTL_IN_EN) { unsigned int cap_set, cap_check; const char *vref; @@ -455,11 +463,27 @@ static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, if (!cap_set) cap_set = AC_PINCAP_VREF_HIZ; if (cap_check && !(cap_set & cap_check)) { - hda_log(HDA_LOG_ERR, - "setting VREF %s to pin 0x%x without caps 0x%x\n", - vref, node->nid, node->pincap); + if (log) + hda_log(HDA_LOG_ERR, + "setting VREF %s to pin 0x%x without caps 0x%x\n", + vref, node->nid, node->pincap); + pinctl &= ~AC_PINCTL_VREFEN; } } + + if (sanified_pinctl) + *sanified_pinctl = pinctl; +} + +static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, + unsigned int cmd) +{ + if (!node) + return 0; + node->pinctl = cmd & 0xff; + + hda_verify_pin_ctl(node, 1, NULL); + return 0; } diff --git a/hda-parse.c b/hda-parse.c index c75b03b..d38bc04 100644 --- a/hda-parse.c +++ b/hda-parse.c @@ -240,6 +240,8 @@ static int parse_node_items(const char *buf) &node->orig_amp_out_vals); } else if ((p = strmatch(head, "Pin-ctls: "))) { node->pinctl = strtoul(p, NULL, 0); + if (node->pincap) + hda_verify_pin_ctl(node, 0, &node->pinctl); node->orig_pinctl = node->pinctl; } else if ((p = strmatch(head, "Pincap "))) { const char *end; diff --git a/include/hda-types.h b/include/hda-types.h index 3754bca..68cfaa8 100644 --- a/include/hda-types.h +++ b/include/hda-types.h @@ -146,6 +146,10 @@ void hda_free_route_lists(struct xhda_route_list *list); void hda_show_routes(int nid, unsigned int flags); +void hda_verify_pin_ctl(struct xhda_node *node, int log, + unsigned int *sanified_pinctl); + + void *xalloc(size_t size); #endif /* __HDA_TYPES_H */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: compress: Only mute playback streams
by Mark Brown 07 Feb '13

07 Feb '13

Otherwise capture activity on a compressed DAI would mute any playback on the same DAI. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index c81aeec..35726cb 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,13 +116,12 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; + snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } - snd_soc_dai_digital_mute(codec_dai, 1); - cpu_dai->active--; codec_dai->active--; codec->active--; @@ -179,10 +178,16 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cmd == SNDRV_PCM_TRIGGER_START) - snd_soc_dai_digital_mute(codec_dai, 0); - else if (cmd == SNDRV_PCM_TRIGGER_STOP) - snd_soc_dai_digital_mute(codec_dai, 1); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1); + break; + } + } out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: core: Allow digital mute for capture
by Mark Brown 07 Feb '13

07 Feb '13

Help avoid noise from the power up of the capture path propagating through into the start of the recording (especially noise caused by the ramp of microphone biases) by keeping the capture muted until after we've finished powering things up with DAPM in the same manner we do for playback. This allows us to take advantage of soft mute support in the hardware more effectively and is more consistent. The core code using the existing digital mute operation is updated to take advantage of this. Some additional cases in the soc-pcm code and suspend will need separate handling but these are less practically relevant than the main runtime stream start/stop case. Rather than refactor the digital mute function in every single driver a new operation is added for drivers taking advantage of this functionality, the old operation should be phased out over time. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 3 ++- sound/soc/soc-compress.c | 19 +++++++++---------- sound/soc/soc-core.c | 12 ++++++++++-- sound/soc/soc-dapm.c | 6 ++++-- sound/soc/soc-pcm.c | 7 +++---- 5 files changed, 28 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 8bcdd71..6afdaf6 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -126,7 +126,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction); struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 35726cb..b5b3db7 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,12 +116,13 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; - snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + cpu_dai->active--; codec_dai->active--; codec->active--; @@ -178,15 +179,13 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - snd_soc_dai_digital_mute(codec_dai, 0); - break; - case SNDRV_PCM_TRIGGER_STOP: - snd_soc_dai_digital_mute(codec_dai, 1); - break; - } + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0, cstream->direction); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + break; } out: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcee552..1283d86 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3547,12 +3547,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); * snd_soc_dai_digital_mute - configure DAI system or master clock. * @dai: DAI * @mute: mute enable + * @direction: stream to mute * * Mutes the DAI DAC. */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) { - if (dai->driver && dai->driver->ops->digital_mute) + if (!dai->driver) + return -ENOTSUPP; + + if (dai->driver->ops->mute_stream) + return dai->driver->ops->mute_stream(dai, mute, direction); + else if (direction == SNDRV_PCM_STREAM_PLAYBACK && + dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else return -ENOTSUPP; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 17dba18..1dc5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3255,14 +3255,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0); + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); ret = 0; break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1); + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); ret = 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 786c2e7..a8d72a9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -383,8 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -488,7 +487,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); out: mutex_unlock(&rtd->pcm_mutex); @@ -586,7 +585,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) -- 1.7.10.4

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Re: [alsa-devel] [PATCH 001/002] ASoC: Replace max98090 Device Driver
by Mark Brown 07 Feb '13

07 Feb '13

On Wed, Feb 06, 2013 at 11:02:33AM -0800, Jerry Wong wrote: > From: Jerry Wong <jerry.wong(a)maximintegrated.com> > > This patch removes the existing max98090 driver prior to installing a more > complete one. Applied both, thanks.

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Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Takashi Iwai 07 Feb '13

07 Feb '13

At Thu, 7 Feb 2013 11:37:38 +0000, Richard Fitzgerald wrote: > > > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > > > 2) Send SNDRV_COMPRESS_DRAIN > > The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which > > expects the decoder to completely drain its buffers and come to complete halt. > > This would also mean the framework will treat a drained stream as stopped and > > needs a new start. Certainly we dont want that in this case. So we can't use > > SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that > > would overtly complicate this. If we are not doing proper drian lets not > > call it that. > > Ok, so let's keep the partial drain but split out the NEXT_TRACK hint so we keep > the two operations separate. It's clearer to understand. NEXT_TRACK tells DSP to > prepare for data for new track, PARTIAL_DRAIN asks DSP when it has played all > data of current track. I don't think that application _must_ drain when doing > gapless, only if it cares to know when DSP reaches the join between tracks. > Possible problem - application calls NEXT_TRACK and PARTIAL_DRAIN when DSP has > already reached end of current track, so what does PARTIAL_DRAIN mean in this > case? Which track is it draining? Maybe define that it always refers to the > track _before_ the most recent NEXT_TRACK call? > > Also can we make metadata a key-pair list as Takashi suggested, with unlimited > length. Something like > > struct snd_compr_metadata_pair { > enum snd_compr_metadata_key key; > u32 value; > }; > > struct snd_compr_metadata { > int count; /* number of actual entries in following array */ > struct snd_compr_metadata_pair *pairs; > }; > > Should we union the u32 value with a u8* buf to allow for any case where the > value needs to be a buffer of binary data? Oh no, don't do it. Never. In general, - Never put types that depending on the architecture in struct passed to ioctls (e.g. long, pointer) - Never use bit fields for ABI - If 64bit value is used, think of alignment; safer to add packed attribute So, in general, use only u32, s16, or whatever, the types with explicit size. Takashi

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Re: [alsa-devel] mmotm 2013-02-06-17-13 uploaded (alsa)
by Takashi Iwai 07 Feb '13

07 Feb '13

At Wed, 06 Feb 2013 21:59:37 -0800, Randy Dunlap wrote: > > On 02/06/13 17:14, akpm(a)linux-foundation.org wrote: > > The mm-of-the-moment snapshot 2013-02-06-17-13 has been uploaded to > > > > http://www.ozlabs.org/~akpm/mmotm/ > > > > mmotm-readme.txt says > > > > README for mm-of-the-moment: > > > > http://www.ozlabs.org/~akpm/mmotm/ > > > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > > more than once a week. > > > > on i386: (from linux-next.patch) > > > CC [M] sound/pci/hda/patch_via.o > sound/pci/hda/patch_via.c: In function 'is_aa_path_mute': > sound/pci/hda/patch_via.c:360:27: error: 'struct hda_gen_spec' has no member named 'num_loopbacks' > sound/pci/hda/patch_via.c:361:17: error: 'struct hda_gen_spec' has no member named 'loopback_list' > make[4]: *** [sound/pci/hda/patch_via.o] Error 1 > > > > Full randconfig file is attached. Oh, it's again an unsatisfied CONFIG_PM ifdef. Fixed now. Thanks! Takashi

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[alsa-devel] [patch] ALSA: ice1712: fix boundary check in snd_wm8766_write()
by Dan Carpenter 07 Feb '13

07 Feb '13

The wm->regs[] array has WM8766_REG_COUNT (16) elements not WM8766_REG_RESET (31). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This is a static checker thing, I can't test it. diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index 8072ade..e473f8a 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -31,7 +31,7 @@ static void snd_wm8766_write(struct snd_wm8766 *wm, u16 addr, u16 data) { - if (addr < WM8766_REG_RESET) + if (addr < WM8766_REG_COUNT) wm->regs[addr] = data; wm->ops.write(wm, addr, data); }

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[alsa-devel] [PATCH] ASoC: arizona: Fix debug logging level for FLLs and AIFs
by Mark Brown 06 Feb '13

06 Feb '13

Use _dbg for debug messages. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ba3bd03..5e77a60 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -57,14 +57,14 @@ #define arizona_fll_warn(_fll, fmt, ...) \ dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_dbg(_fll, fmt, ...) \ - dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + dev_dbg(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_aif_err(_dai, fmt, ...) \ dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_warn(_dai, fmt, ...) \ dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_dbg(_dai, fmt, ...) \ - dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, -- 1.7.10.4

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