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

[alsa-devel] [PATCH] ASoC: wm8978: enable symmetric rates
by Kuninori Morimoto 12 Jul '13

12 Jul '13

wm8978 needs .symmetric_rates = 1. The playback/capture will be strange without this patch when it used asymmetric rate in same time Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8978.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 029f31c..d8fc531 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -921,6 +921,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .formats = WM8978_FORMATS, }, .ops = &wm8978_dai_ops, + .symmetric_rates = 1, }; static int wm8978_suspend(struct snd_soc_codec *codec) -- 1.7.9.5

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[alsa-devel] [PATCH 0/5] ASoC: add simple-card DT support
by Kuninori Morimoto 12 Jul '13

12 Jul '13

Hi Mark, Simon These patches add DT support on simple-card driver. #1 patch adds new snd_soc_of_parse_daifmt() function on soc-core.c and #5 patch will use it. #2 - #5 patches are based on mark/topic/fsi branch >> Simon #2, #3 patches modify sh-arm platform settings. please check it Kuninori Morimoto (5): ASoC: add snd_soc_of_parse_daifmt() ASoC: simple-card: use struct device pointer for dev_xxx() ASoC: simple-card: add asoc_simple_dai_set for initializing ASoC: simple-card: remove pointless struct asoc_simple_dai_init_info ASoC: simple-card: add Device Tree support .../devicetree/bindings/sound/simple-card.txt | 53 ++++++++ arch/arm/mach-shmobile/board-ap4evb.c | 19 +-- arch/arm/mach-shmobile/board-armadillo800eva.c | 18 +-- arch/arm/mach-shmobile/board-kzm9g.c | 12 +- arch/arm/mach-shmobile/board-mackerel.c | 19 +-- arch/sh/boards/mach-ecovec24/setup.c | 11 +- arch/sh/boards/mach-se/7724/setup.c | 13 +- include/sound/simple_card.h | 11 +- include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 132 ++++++++++++++++---- sound/soc/soc-core.c | 115 +++++++++++++++++ 11 files changed, 307 insertions(+), 98 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v2 0/4] ASoC: Regression fix for OMAP in DT boot (3.11)
by Peter Ujfalusi 11 Jul '13

11 Jul '13

Hi, Changes since v1: - Drop the change in dmaengine_pcm.h - Use .filter_data to store the DMA channel name - omap-pcm: checks if the DAI is probed via DT and uses decides which method to use to request the DMA channel - Dropped the cleanup patch for omap-dmic from the series I will send another series with cleanups later I just noticed in mainline kernel (pre 3.11-rc1) that OMAP4 audio is not probing anymore. This is because OMAP4 also moved to DT only mode (3.10 DT boot was fine). The issue underneath is that we can not use platform_get_resource_byname() for DMA resources when booted with devicetree. In case of DT boot we can use the DMA name to request the channel. Due to technical issues OMAP can not be moved to generic dmaengine pcm as other SoCs did. Mark: can you queue this series for 3.11? Thank you. Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-pcm: Request the DMA channel differently when DT is involved ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 11 +---------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 17 ++++++++++++++--- 4 files changed, 39 insertions(+), 44 deletions(-) -- 1.8.2.1

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[alsa-devel] [PATCH 1/1] ASoC: Samsung: Fix a trivial typo in audss documentation
by Sachin Kamat 11 Jul '13

11 Jul '13

Fixed a trivial typo. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- .../devicetree/bindings/clock/clk-exynos-audss.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt index a120180..75e2e19 100644 --- a/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt +++ b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt @@ -2,7 +2,7 @@ The Samsung Audio Subsystem clock controller generates and supplies clocks to Audio Subsystem block available in the S5PV210 and Exynos SoCs. The clock -binding described here is applicable to all SoC's in Exynos family. +binding described here is applicable to all SoCs in Exynos family. Required Properties: -- 1.7.9.5

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Re: [alsa-devel] [PATCH 45/50] sound: usb: usx2y: spin_lock in complete() cleanup
by Takashi Iwai 11 Jul '13

11 Jul '13

At Thu, 11 Jul 2013 22:13:35 +0800, Ming Lei wrote: > > On Thu, Jul 11, 2013 at 9:50 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 11 Jul 2013 17:08:30 +0400, > > Sergei Shtylyov wrote: > >> > >> On 11-07-2013 13:06, Ming Lei wrote: > >> > >> > Complete() will be run with interrupt enabled, so change to > >> > spin_lock_irqsave(). > >> > >> Changelog doesn't match the patch. > > > > Yep, but moreover... > > > >> > Cc: Jaroslav Kysela <perex(a)perex.cz> > >> > Cc: Takashi Iwai <tiwai(a)suse.de> > >> > Cc: alsa-devel(a)alsa-project.org > >> > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > >> > --- > >> > sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ > >> > 1 file changed, 4 insertions(+) > >> > >> > diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c > >> > index 4967fe9..e2ee893 100644 > >> > --- a/sound/usb/usx2y/usbusx2yaudio.c > >> > +++ b/sound/usb/usx2y/usbusx2yaudio.c > >> > @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) > >> > struct snd_usX2Y_substream *subs = usX2Y->subs[s]; > >> > if (subs) { > >> > if (atomic_read(&subs->state) >= state_PRERUNNING) { > >> > + unsigned long flags; > >> > + > >> > + local_irq_save(flags); > >> > snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); > >> > + local_irq_restore(flags); > >> > } > > > > ... actually this snd_pcm_stop() call should have been covered by > > snd_pcm_stream_lock(). Maybe it'd be enough to have a single patch > > together with the change, i.e. wrapping with > > snd_pcm_stream_lock_irqsave(). > > Please use snd_pcm_stream_lock_irqsave() so that I can avoid sending > out similar patch later, :-) > > > > > I'll prepare the patch for 3.11 independently from your patch series, > > so please drop this one. > > OK, thanks for dealing with that. > > > > > > > BTW, the word "cleanup" in the subject is inappropriate. This is > > rather a fix together with the core change. > > It is a cleanup since the patchset only addresses lock problem which > is caused by the tasklet conversion. Well, the conversion to irqsave() is needed for the future drop of irq_save() in the caller, right? Then this isn't a cleanup but a preparation for movement ahead. > > And, I wonder whether we can take a safer approach. When the caller > > condition changed, we often introduced either a different ops > > (e.g. ioctl case) or a flag for the new condition, at least during the > > transition period. > > Interrupt is't enabled until all current drivers are cleaned up, so it is really > safe, please see patch [2]. OK. > > Last but not least, is a conversion to tasklet really preferred? > > tasklet is rather an obsoleted infrastructure nowadays, and people > > don't recommend to use it any longer, AFAIK. > > We discussed the problem in the below link previously[1], Steven > and Thomas suggested to use threaded irq handler, but which > may degrade USB mass storage performance, so we have to > take tasklet now until we rewrite transport part of USB mass storage > driver. > > Also the conversion[2] has avoided the tasklet spin lock problem > already. OK, good to know. thanks, Takashi > [1], http://marc.info/?t=137079119200001&r=1&w=2 > [2], http://marc.info/?l=linux-usb&m=137286326726326&w=2 > > Thanks, > -- > Ming Lei >

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Re: [alsa-devel] [PATCH 44/50] sound: usb: caiaq: spin_lock in complete() cleanup
by Daniel Mack 11 Jul '13

11 Jul '13

On 11.07.2013 11:06, Ming Lei wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Daniel Mack <zonque(a)gmail.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> Sound right to me, thanks. Acked-by: Daniel Mack <zonque(a)gmail.com> > --- > sound/usb/caiaq/audio.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c > index 7103b09..e5675ab 100644 > --- a/sound/usb/caiaq/audio.c > +++ b/sound/usb/caiaq/audio.c > @@ -672,10 +672,11 @@ static void read_completed(struct urb *urb) > offset += len; > > if (len > 0) { > - spin_lock(&cdev->spinlock); > + unsigned long flags; > + spin_lock_irqsave(&cdev->spinlock, flags); > fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); > read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); > - spin_unlock(&cdev->spinlock); > + spin_unlock_irqrestore(&cdev->spinlock, flags); > check_for_elapsed_periods(cdev, cdev->sub_playback); > check_for_elapsed_periods(cdev, cdev->sub_capture); > send_it = 1; >

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Re: [alsa-devel] [PATCH 45/50] sound: usb: usx2y: spin_lock in complete() cleanup
by Takashi Iwai 11 Jul '13

11 Jul '13

At Thu, 11 Jul 2013 17:08:30 +0400, Sergei Shtylyov wrote: > > On 11-07-2013 13:06, Ming Lei wrote: > > > Complete() will be run with interrupt enabled, so change to > > spin_lock_irqsave(). > > Changelog doesn't match the patch. Yep, but moreover... > > Cc: Jaroslav Kysela <perex(a)perex.cz> > > Cc: Takashi Iwai <tiwai(a)suse.de> > > Cc: alsa-devel(a)alsa-project.org > > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > > --- > > sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ > > 1 file changed, 4 insertions(+) > > > diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c > > index 4967fe9..e2ee893 100644 > > --- a/sound/usb/usx2y/usbusx2yaudio.c > > +++ b/sound/usb/usx2y/usbusx2yaudio.c > > @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) > > struct snd_usX2Y_substream *subs = usX2Y->subs[s]; > > if (subs) { > > if (atomic_read(&subs->state) >= state_PRERUNNING) { > > + unsigned long flags; > > + > > + local_irq_save(flags); > > snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); > > + local_irq_restore(flags); > > } ... actually this snd_pcm_stop() call should have been covered by snd_pcm_stream_lock(). Maybe it'd be enough to have a single patch together with the change, i.e. wrapping with snd_pcm_stream_lock_irqsave(). I'll prepare the patch for 3.11 independently from your patch series, so please drop this one. BTW, the word "cleanup" in the subject is inappropriate. This is rather a fix together with the core change. And, I wonder whether we can take a safer approach. When the caller condition changed, we often introduced either a different ops (e.g. ioctl case) or a flag for the new condition, at least during the transition period. Last but not least, is a conversion to tasklet really preferred? tasklet is rather an obsoleted infrastructure nowadays, and people don't recommend to use it any longer, AFAIK. thanks, Takashi

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Re: [alsa-devel] [PATCH 36/50] media: usb: em28xx: spin_lock in complete() cleanup
by Devin Heitmueller 11 Jul '13

11 Jul '13

On Thu, Jul 11, 2013 at 5:05 AM, Ming Lei <ming.lei(a)canonical.com> wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: linux-media(a)vger.kernel.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > drivers/media/usb/em28xx/em28xx-core.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/drivers/media/usb/em28xx/em28xx-core.c b/drivers/media/usb/em28xx/em28xx-core.c > index fc157af..0d698f9 100644 > --- a/drivers/media/usb/em28xx/em28xx-core.c > +++ b/drivers/media/usb/em28xx/em28xx-core.c > @@ -941,6 +941,7 @@ static void em28xx_irq_callback(struct urb *urb) > { > struct em28xx *dev = urb->context; > int i; > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -956,9 +957,9 @@ static void em28xx_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->slock); > + spin_lock_irqsave(&dev->slock, flags); > dev->usb_ctl.urb_data_copy(dev, urb); > - spin_unlock(&dev->slock); > + spin_unlock_irqrestore(&dev->slock, flags); > > /* Reset urb buffers */ > for (i = 0; i < urb->number_of_packets; i++) { > -- > 1.7.9.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-media" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html I actually stumbled across this a couple of weeks ago, and have had an identical patch running in a local dev tree. Reviewed-by: Devin Heitmueller <dheitmueller(a)kernellabs.com> Tested-by: Devin Heitmueller <dheitmueller(a)kernellabs.com> -- Devin J. Heitmueller - Kernel Labs http://www.kernellabs.com

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[alsa-devel] [PATCH 0/6] ASoC: Regression fix for OMAP in DT boot (3.11)
by Peter Ujfalusi 11 Jul '13

11 Jul '13

Hi, I just noticed in mainline kernel (pre 3.11-rc1) that OMAP4 audio is not probing anymore. This is because OMAP4 also moved to DT only mode (3.10 DT boot was fine). The issue underneath is that we can not use platform_get_resource_byname() for DMA resources when booted with devicetree. In case of DT boot we can use the DMA name to request the channel. Due to technical issues OMAP can not be moved to generic dmaengine pcm as other SoCs did. Mark: can you queue this series for 3.11? Thank you. Regards, Peter --- Peter Ujfalusi (6): ASoC: dmaengine_pcm: Add dma_name to snd_dmaengine_dai_dma_data ASoC: omap-pcm: Request the DMA channel by name when dma_name is set for the dai ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT ASoC: omap-dmic: Cleanup the probe function include/sound/dmaengine_pcm.h | 1 + sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 35 ++++++++++------------------------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 16 +++++++++++++--- 5 files changed, 48 insertions(+), 59 deletions(-) -- 1.8.2.1

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[alsa-devel] Problem with MX27 ASoC audio
by Randolph Chung 11 Jul '13

11 Jul '13

I have a target that uses iMX27(L) + tlv320aic3x. The kernel tree is HEAD of linux-arm from about a week ago. Simple playback and recording (using aplay and arecord) work fine. However, I've run into a strange problem when both playback and capture streams are opened at the same time. Consider the attached test program. The capture device is setup with a period time of 10ms, and the program reads samples in a loop. The expected result is that I get a sample every 10ms. This works fine if the playback device is *not* opened (more specifically, if the hw_params are not set). However if the playback device is opened (#if 1 code) then I get samples every 20ms, which seems very wrong. I have tried using both DMA and FIQ ssi-pcm interfaces and both give the same strange result. With the FIQ driver, if I instrument the timer callback, the timer triggers every 10ms but there is only enough data every 20ms. The codec is being driven with a 13MHz MCLK (output from MX27) and the BCLK and WCLK come from the codec. They seem to have the correct rates. Any ideas? Can anybody else with MX27 hardware verify if my test program also fails on their target? thanks, randolph /* Test program */ #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #include <assert.h> #include <sys/time.h> int main(int argc, char **argv) { int err = -1; char buf[4096]; unsigned long period_size; snd_pcm_t *in_handle = NULL, *out_handle; snd_pcm_hw_params_t *hwparams; if (snd_pcm_open(&in_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) < 0) { printf("Cannot open capture device\n"); return -1; } snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(in_handle, hwparams); snd_pcm_hw_params_set_access(in_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(in_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(in_handle, hwparams, 1); snd_pcm_hw_params_set_rate(in_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(in_handle, hwparams, 10000, 0); snd_pcm_hw_params_get_period_size(hwparams, &period_size, 0); snd_pcm_hw_params(in_handle, hwparams); snd_pcm_hw_params_free(hwparams); if (snd_pcm_open(&out_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0) < 0) { printf("Cannot open playback device\n"); return -1; } #if 1 snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(out_handle, hwparams); snd_pcm_hw_params_set_access(out_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(out_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(out_handle, hwparams, 1); snd_pcm_hw_params_set_rate(out_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(out_handle, hwparams, 10000, 0); snd_pcm_hw_params(out_handle, hwparams); snd_pcm_hw_params_free(hwparams); #endif printf("period size = %u\n", period_size); printf("Config done, start loop\n"); while (1) { struct timeval tv; int ret = snd_pcm_readi(in_handle, buf, period_size); if (ret < 0) snd_pcm_prepare(in_handle); gettimeofday(&tv, NULL); printf("[%u.%06u] snd_pcm_readi() return %d\n", tv.tv_sec, tv.tv_usec, ret); } return 0; }

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