Alsa-devel September 2013

alsa-devel@alsa-project.org

119 participants
213 discussions

[alsa-devel] [PATCH v3 00/11] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 01 Sep '13

01 Sep '13

Hi, This patch series attempts to fix a long standing problem with the concurrent usage of playback and capture on implicit feedback devices. Applies against current for-next (HEAD 68538bf2bce557c3b5fe8c59b034d45352500db1 ). v2: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065649.html v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065517.html * Changed "altset_idx" to "altsetting" thanks to Clemens' clarification. #3: Thanks to Daniel's test, fixed a NULL dereference caused by userspace calling snd_pcm_open() and then snd_pcm_close() immediately. #8: subs->altset_idx was left intact, since it corresponds to the array index. #10: this patch now removed subs->altset_idx, but also changes the check to be performed against fmt->altsetting. #11: New. This changes the name of snd_usb_endpoint member alt_idx to altsetting for the same reason Clemens described. Clemens and Daniel - thank you very much for your help! Cheers, Eldad Eldad Zack (11): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: correct ep use_count semantics (add set_param flag) ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: remove altset_idx from snd_usb_substream ALSA: usb-audio: rename alt_idx to altsetting sound/usb/card.h | 4 +- sound/usb/endpoint.c | 82 ++++++++++++++++++------- sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 168 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/pcm.h | 2 + sound/usb/proc.c | 4 +- 6 files changed, 201 insertions(+), 70 deletions(-) -- 1.8.1.5

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Re: [alsa-devel] [PATCH 2/2] ARM: Dove: Add the audio device to the Cubox DT
by Jean-Francois Moine 01 Sep '13

01 Sep '13

On Sat, 31 Aug 2013 12:24:31 +0100 Russell King - ARM Linux <linux(a)arm.linux.org.uk> wrote: > > According to the Dove specification, the audio controller works with > > the samples rates 44.1, 48 and 96 kHz, so, I don't see the usage of the > > external clock, except when using the two audio controllers with > > different sample rates. > > I don't see what the Dove specification has to do with that statement: > what the Dove spec says is that if you use just the internal DCO, then > only 44.1kHz, 48kHz and 96kHz are supported (with some trimming of that.) > However, the use of an external clock allows further rates to be supported. > If you have an external clock, there is no requirement to use the DCO for > those sample rates - you can if you wish, or you can use the external clock. > > The mainline driver implements the use of the DCO for the standard 44.1, > 48 and 96kHz rates, otherwise it uses the external clock if present. This > is entirely conformant with the Dove spec. Russell, I looked again at the Armada-510-Functional-Spec, and I found: - section 12.1 ... I2S Supported Features An audio sample rate (Fs) of 44.1/48/96 kHz. ... S/PDIF Supported Features The IEC60958-1, IEC60958-3, and IEC61937 specifications. An audio sample rate (Fs) of 44.1/48/96 kHz. ... - Table 436: Clocks Control Register Offset: Unit0: 0x000B1230, Unit1: 0x000B5230 ... 1:0 MCLK source RW MCLK = 256 Fs, where Fs = 44.1 kHz, 48 kHz, 96 kHz. 0x0 Playback should not be active when selecting an external MCLK, or moving from one external MCLK to another MCLK source. 0 = DCO: MCLK is from DCO. 1 = Reserved 2 = Reserved 3 = External: MCLK is from an external source. As they don't work with my TV set, may you confirm that these webradios http://kerne.online.stalig.net/live-ori.ogg http://www.tv-radio.com/station/france_bleu_breiz_izel_mp3/france_bleu_brei… http://plumfm.online.stalig.net/live-ori.ogg work for you with the external clock? (sample rates resp. 33.075, 24 and 22.05 kHz) -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH] ASoC: fsl_spdif: Staticse non-exported symbols
by Mark Brown 01 Sep '13

01 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..c5c228a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, return 0; } -int fsl_spdif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int fsl_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); @@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, return 0; } -struct snd_soc_dai_ops fsl_spdif_dai_ops = { +static struct snd_soc_dai_ops fsl_spdif_dai_ops = { .startup = fsl_spdif_startup, .hw_params = fsl_spdif_hw_params, .trigger = fsl_spdif_trigger, @@ -919,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) return 0; } -struct snd_soc_dai_driver fsl_spdif_dai = { +static struct snd_soc_dai_driver fsl_spdif_dai = { .probe = &fsl_spdif_dai_probe, .playback = { .channels_min = 2, -- 1.8.4.rc3

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Alsa-devel September 2013