September 2011 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 3/4] ASoC: Staticise jz4740_pcm_new()
by Lars-Peter Clausen 21 Sep '11

21 Sep '11

On 09/21/2011 08:41 AM, Axel Lin wrote: > It is not used outside this driver so no need to make the symbol global. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Thanks. > --- > sound/soc/jz4740/jz4740-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c > index a7c9578..d1989cd 100644 > --- a/sound/soc/jz4740/jz4740-pcm.c > +++ b/sound/soc/jz4740/jz4740-pcm.c > @@ -299,7 +299,7 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) > > static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); > > -int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) > +static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_soc_dai *dai = rtd->cpu_dai;

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Re: [alsa-devel] [PATCH 1/4] ASoC: Staticise bf5xx_pcm_ac97_new()
by Mark Brown 21 Sep '11

21 Sep '11

On Wed, Sep 21, 2011 at 02:40:05PM +0800, Axel Lin wrote: > It is not used outside this driver so no need to make the symbol global. All applied, thanks.

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[alsa-devel] Using the timer
by Ilya Dmitrichenko 21 Sep '11

21 Sep '11

Hi, I would like to utilise the hardware timer which is in the soundcard in an application. I suppose that I can somehow set snd-seq-timer to use a particular device as master and access that via sequencer API? The particular device that I have is Hammerfall HDSP (snd-hdsp). Cheers, -- Ilya

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[alsa-devel] [PATCH 00/17] ASoC: twl6040: Cleanups and fixes
by Peter Ujfalusi 21 Sep '11

21 Sep '11

Hello, The first four patch of this series (the MFD/Input patches) has been already sent to upstream to the relevant lists, but I wanted to include here as well, since I have dependency on that series later on. If it is OK with you, it would be best to merge these MFD/Input changes through the audio tree for 3.2. Code cleanup, and fixes for the twl6040 codec driver. Main changes: - reg_cache has been rested to chip defaults - codec initialization changes - Correction for the Earphone path - AUX L/R output support - Headset/Handsfree control name changes - Headset DC offset workaround Regards, Peter I have excluded the linux-input list (but kept Dmitry) - to not spam them. --- Peter Ujfalusi (17): MFD: twl6040: Remove global pointer for platform_device MFD: twl6040: Add accessor for revision ID Input: twl6040-vibra: Use accessor to get revision information MFD: twl6040: Fix power on GPIO handling ASoC: twl6040: Chip initialization cleanup ASoC: twl6040: Use chip defaults in the initial reg_cache ASoC: twl6040: Lower the initial gains ASoC: twl6040: Fix comments for register names ASoC: twl6040: Remove strings "NULL" from DAPM route ASoC: twl6040: Introduce SW only shadow register ASoC: twl6040: Earphone path correction ASoC: twl6040: Use consistent names for Handsfree path ASoC: twl6040: Use consistent names for Headset path ASoC: twl6040: Support for AUX L/R output ASoC: twl6040/sdp4430: Change legacy DAI name ASoC/MFD: twl6040: Combine bit definitions for Headset control registers ASoC: twl6040: Workaround for headset DC offset caused pop noise drivers/input/misc/twl6040-vibra.c | 2 +- drivers/mfd/twl6040-core.c | 38 ++-- include/linux/mfd/twl6040.h | 22 +-- sound/soc/codecs/twl6040.c | 367 +++++++++++++++++++----------------- sound/soc/omap/sdp4430.c | 2 +- 5 files changed, 218 insertions(+), 213 deletions(-) -- 1.7.6.1

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[alsa-devel] [PATCH] ALSA: snd-aloop: make preallocated buffer size configurable
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Default of 2MB of physically-contiguous memory can be too high for embedded solutions, make the size configurable. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/drivers/Kconfig | 15 +++++++++++++++ sound/drivers/aloop.c | 6 +++--- 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index c896116..ad51067 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -93,6 +93,21 @@ config SND_ALOOP To compile this driver as a module, choose M here: the module will be called snd-aloop. +if SND_ALOOP +config SND_ALOOP_PREALLOC_SIZE + int "Pre-allocated buffer size for aloop driver" + range 0 32768 + default 2048 + help + Specifies the default pre-allocated buffer-size in kB for the + snd-aloop driver. The default 2048 is chosen just + for compatibility reasons. Some distributions may want to + use a smaller size when only one stream is used. + + Note that the pre-allocation size can be changed dynamically + via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. +endif + config SND_VIRMIDI tristate "Virtual MIDI soundcard" depends on SND_SEQUENCER diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a0da775..f945573 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -556,11 +556,11 @@ static struct snd_pcm_hardware loopback_pcm_hardware = .rate_max = 192000, .channels_min = 1, .channels_max = 32, - .buffer_bytes_max = 2 * 1024 * 1024, + .buffer_bytes_max = CONFIG_SND_ALOOP_PREALLOC_SIZE * 1024, .period_bytes_min = 64, /* note check overflow in frac_pos() using pcm_rate_shift before changing period_bytes_max value */ - .period_bytes_max = 1024 * 1024, + .period_bytes_max = CONFIG_SND_ALOOP_PREALLOC_SIZE * 1024, .periods_min = 1, .periods_max = 1024, .fifo_size = 0, @@ -774,7 +774,7 @@ static int __devinit loopback_pcm_new(struct loopback *loopback, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), - 0, 2 * 1024 * 1024); + 0, CONFIG_SND_ALOOP_PREALLOC_SIZE * 1024); return 0; } -- 1.7.6

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[alsa-devel] link between HDMI ELD and PCM devices
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Hi, I've been trying for some time to expose the HDMI ELD information in a new control. The problem I am facing with the HDMI/HDA code is that the ELD is linked to an output pin, and I can't find an elegant way to link this output pin with a PCM device number to be used by my control. For a single output, I could hard-code something but it would probably not work with multiple monitors handled by recent hardware. There is some existing code such as pin_nid_to_pin_index() in patch_hdmi.c, but I can't figure out what this pin_idx is, and how to find the related PCM device number. Any suggestions? Thanks, -Pierre

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Re: [alsa-devel] Questions for dummy codec driver
by Mark Brown 21 Sep '11

21 Sep '11

On Wed, Sep 21, 2011 at 10:23:18AM +0800, Leo Yan wrote: > I have a question for the dummy codec. Don't start new threads by randomly replying to unrelated mails. > As you know, sometimes maybe we only need set the timing from the > processor, and codec just only need tx/rx data; So there do not need > the codec driver. Even a dumb CODEC will have some constraints on things like the sample rates it can support which need to be conveyed to the framework and ideally the audio routing out of the CODEC should be visible for integration with any external output drivers or similar. > After browser the code, there have NO such a dummy codec driver; and > for soc-core, there have not such flags to support this situation. > Only what i get is sound/soc/codecs/spdif_transciever.c, but it's > specific for TI's DaVinci platform. Not at all, what makes you believe that it is platform specific?

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[alsa-devel] [PATCH] ASoC: Ensure we generate a driver name
by Mark Brown 21 Sep '11

21 Sep '11

Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver field) broke generation of a driver name for all ASoC cards relying on the automatic generation of one. Fix this by using the old default with spaces replaced by underscores. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 10e5cde..bd20154 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -30,6 +30,7 @@ #include <linux/bitops.h> #include <linux/debugfs.h> #include <linux/platform_device.h> +#include <linux/ctype.h> #include <linux/slab.h> #include <sound/ac97_codec.h> #include <sound/core.h> @@ -1432,9 +1433,20 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); - if (card->driver_name) - strlcpy(card->snd_card->driver, card->driver_name, - sizeof(card->snd_card->driver)); + snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), + "%s", card->driver_name ? card->driver_name : card->name); + for (i = 0; i < ARRAY_SIZE(card->snd_card->driver); i++) { + switch (card->snd_card->driver[i]) { + case '_': + case '-': + case '\0': + break; + default: + if (!isalnum(card->snd_card->driver[i])) + card->snd_card->driver[i] = '_'; + break; + } + } if (card->late_probe) { ret = card->late_probe(card); -- 1.7.6.3

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Re: [alsa-devel] [PATCH] ASoC: Flush Samsung DMA on free
by Kukjin Kim 21 Sep '11

21 Sep '11

Boojin Kim wrote: > > > Mark Brown wrote: > > > > Ever since it was written the Samsung DMA driver has had a TODO in the > > hw_free() function wondering if we need to flush the DMA buffers. Up > > until > > now the answer has been no but with the recent improvements Boojin has > > done to the DMA infrastructure for the Samsung port the answer has > > changed > > to yes for at least S3C6410 systems. > > > > If we don't then when we next prepare() the channel the API will get > > confused trying to run callbacks on the transfers hanging around from > > the > > previous time the stream was open and oops. > > > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > > Your patch give good effect to all Samsung SoCs. > Thanks a lot. > > Acked-by: Boojin Kim <boojin.kim(a)samsung.com> > > > --- > > > > Kukjin, this should probably get added to your branch assuming it's OK > > for other CPUs. > > Looks ok to me, will apply. Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd. > > sound/soc/samsung/dma.c | 2 +- > > 1 files changed, 1 insertions(+), 1 deletions(-) > > > > diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c > > index 851346f..2d622b6 100644 > > --- a/sound/soc/samsung/dma.c > > +++ b/sound/soc/samsung/dma.c > > @@ -198,10 +198,10 @@ static int dma_hw_free(struct snd_pcm_substream > > *substream) > > > > pr_debug("Entered %s\n", __func__); > > > > - /* TODO - do we need to ensure DMA flushed */ > > snd_pcm_set_runtime_buffer(substream, NULL); > > > > if (prtd->params) { > > + prtd->params->ops->flush(prtd->params->ch); > > prtd->params->ops->release(prtd->params->ch, > > prtd->params->client); > > prtd->params = NULL; > > -- > > 1.7.6.3

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[alsa-devel] [PATCH] ALSA: hda: hdmi: Hint matching between input devices and pcm devices
by David Henningsson 21 Sep '11

21 Sep '11

Hi Takashi, You went on vacation when this patch was posted and discussed, but would you mind reviewing/committing this patch now that you're back? It was reviewed by Stephen Warren who thought the patch was okay, but had some thoughts on doing other changes to the format of that string at the same time [1]. I had to distro patch a 3.0 adapted version of this patch into Ubuntu in order to meet the Ubuntu 11.10 kernel deadline, so for that reason I'd say that the ",pcm=x" format would be preferrable for me, and while I don't mind further improvements to the string format in the future, this would be a step in the right direction and IMO this patch should not block on such future improvements. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/043258.html

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