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

[alsa-devel] [PATCH v2] ASoC: dapm - Add DAPM stream completion event.
by Liam Girdwood 25 Jul '11

25 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 4 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..5833f01 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -518,6 +518,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..64a9dd5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -634,6 +634,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -669,6 +672,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..159f144 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,6 +3140,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3252,6 +3253,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fbfcda0..80f97b2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2574,6 +2574,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm, event); } /** -- 1.7.4.1

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[alsa-devel] M-Audio FTU issues
by Daniel Mack 25 Jul '11

25 Jul '11

So, I've looked into this today and as expected, the device normally operates in implicit feedback mode. After installing a proprietary driver on Mac OS X (that regularily crashed my kernel), the analyzer shows that for every inbound packet, an out packet is sent back with the same amount of data in it. And there are no feedback endpoints. On Tue, Jun 14, 2011 at 11:21 AM, Aurélien Leblond <blablack(a)gmail.com> wrote: > There are still the issue of samples dropping. > > Tto reproduce it: > - Open Audacity and set it up to be using Jack. > - Click Generate -> Tone -> Ok. > > "Normally" you should hear very regular pops... > What surprises me though, is that I can't hear the clicks and pops you guys describe. Instead, I hear a quite annoying distortion for both 44100 and 48000 Hz which I can't explain either. My spontanous guess is that there is a resampler in the chain which compensates the sample drift and notably reduces the audio quality. I patched the driver so that it supports implicit feedback, but that didn't change anything in that regard. Hence my question is: How do you guys test? I renderened a sine tone in different samplerates and bit depths to .wav files and play them with aplay (no pulseaudio or anything in the chain). Could you try the same and report what happens? Daniel

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[alsa-devel] [PATCH V4 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 25 Jul '11

25 Jul '11

Hello, Here's V4 of the AC97/I2S ASoC patchset for early Alchemy chips and their respective evaluation boards. The patches are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V3: - dropped the hunk which removed the I2S constants from the au1000.h header to avoid merge conflicts with other patches. - use the context structure declared in psc.h. Follow-up patches for psc* code depend on this. Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 ++++ sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 365 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 374 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 342 ++++++++++++++++++++++ sound/soc/au1x/psc.h | 19 +- 9 files changed, 1256 insertions(+), 10 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] ac97: quiet make W=1 noise by initializing all fields of snd_ac97_codec_id_vendors and snd_ac97_codec_ids.
by Takashi Iwai 25 Jul '11

25 Jul '11

At Sun, 24 Jul 2011 11:40:56 +0200, Lionel Debroux wrote: > > make W=1 bitterly complains about missing initializations in > snd_ac97_codec_id_vendors and snd_ac97_codec_ids. > Initialize the last field, to make the output of make W=1 more readable. > > Patch against current Linus tree. If we change the whole table entries, it'd be better to use C99 style init, e.g. #define _CODEC(_id, _mask, _name) \ .id = _id, .mask = _mask, .name = _name static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { { _CODEC(0x41445300, 0xffffff00, "Analog Devices"),}, ... static const struct ac97_codec_id snd_ac97_codec_ids[] = { { _CODEC(0x41445303, 0xffffffff, "AD1819"), .patch = patch_ad1819 }, ... { _CODEC(0x53494c20, 0xffffffe0, "Si3036,8"), .patch = mpatch_si3036, \ .mpatch = mpatch_si3036, .flags = AC97_MODEM_PATCH }, ... thanks, Takashi

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[alsa-devel] [PATCH] ASoC: Add regmap as a control type
by Mark Brown 25 Jul '11

25 Jul '11

Allow drivers to set up their own regmap API structures. This is mainly useful with MFDs where the core driver will have set up regmap at the minute, though it may make sense to push the existing regmap setup out of the core into the drivers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-io.c | 4 ++++ 2 files changed, 5 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4d04b4b..d022694 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -261,6 +261,7 @@ extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { SND_SOC_I2C = 1, SND_SOC_SPI, + SND_SOC_REGMAP, }; enum snd_soc_compress_type { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index b56e1c4..e471ed6 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -132,6 +132,10 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, &config); break; + case SND_SOC_REGMAP: + /* Device has made its own regmap arrangements */ + break; + default: return -EINVAL; } -- 1.7.5.4

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[alsa-devel] [PATCH 0/2] ASoC: au1x: update PSC AC97/I2S code
by Manuel Lauss 25 Jul '11

25 Jul '11

The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work. Patch #1 moves the DBDMA (PCM) device registration from the AC97/I2S drivers into the board code. Patch #2 changes use of "soc-audio" platform device int db1200 machine code to the new way. Run-tested on DB1200, DB1300 and DB1550 boards. Manuel Lauss (2): ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things arch/mips/alchemy/devboards/db1200/platform.c | 16 +++++ sound/soc/au1x/db1200.c | 64 +++++++++++++------ sound/soc/au1x/dbdma2.c | 83 ++++--------------------- sound/soc/au1x/psc-ac97.c | 34 +++++++--- sound/soc/au1x/psc-i2s.c | 32 +++++++--- sound/soc/au1x/psc.h | 5 -- 6 files changed, 117 insertions(+), 117 deletions(-) -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 25 Jul '11

25 Jul '11

The following changes since commit 440085598672c0e3fde8a48495f61fea418b06d1: Merge branch 'for-3.0' into for-3.1 (2011-07-17 18:25:58 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Sorry, I thought I'd sent this out and then the release caught me on the hop a bit. Apart from Liam's changes and the new driver these are all bug fixes of one form or another - the sgtl5000 driver apparently just didn't work most of the time. Liam Girdwood (2): ASoC: dapm - add DAPM macro for external enum widgets ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context. Mark Brown (3): ASoC: Mark cache as dirty when suspending ASoC: Acknowledge WM8962 interrupts before acting on them ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver" Rajashekhara, Sudhakar (2): ASoC: davinci: fix codec start and stop functions ASoC: davinci: add missing break statement Sangbeom Kim (2): ASoC: SAMSUNG: Modify I2S driver to support idma ASoC: SAMSUNG: Add I2S0 internal dma driver Wolfram Sang (3): ASoC: core: make comments fit the code ASoC: sgtl5000: refactor registering internal ldo ASoC: sgtl5000: guide user when regulator support is needed include/sound/soc-dapm.h | 6 +++ sound/soc/codecs/sgtl5000.c | 70 +++++++++++++++++-------------------- sound/soc/codecs/wm8962.c | 6 ++-- sound/soc/davinci/davinci-vcif.c | 9 +++-- sound/soc/samsung/i2s.c | 7 ++++ sound/soc/soc-core.c | 5 ++- sound/soc/soc-dapm.c | 30 ++++++++++++++++ 7 files changed, 86 insertions(+), 47 deletions(-)

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[alsa-devel] lib pcm DMA problems
by Horner, Richard 24 Jul '11

24 Jul '11

To Whom It May Concern: I see some weird behavior from the PCI audio driver for the Hauppauge analog tv card (model 44842, rev D1B2) where the card is tuning properly but audio is no longer available over PCI bus. This condition can happen within an hour or as late as after multiple months of 24x7 operation. Re-loading the TV card and ALSA driver stacks do not clear this condition. Only a warm boot clears it up. Debug prints in the PCI device sound driver, bt87x.c. "snd_bt87x_interrupt()" handler indicate it is returning only IRQ_NONE when in this state. We also get a DMA error message from pcm_lib.c which states "capture write error (DMA or IRQ trouble?)", which I presume means DMA is turned off for this device. How can I recover from this without a system reboot? Thank you so much! Here is my system info: # cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 7 model name : VIA Samuel 2 stepping : 3 cpu MHz : 533.309 cache size : 64 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr cx8 mtrr pge mmx 3dnow up bogomips : 1066.61 clflush size : 32 power management: ----------------------------------------------------------- Interrupts are still active on IRQ 5 during this state: # cat /proc/interrupts CPU0 0: 177526733 XT-PIC-XT timer 1: 2 XT-PIC-XT i8042 2: 0 XT-PIC-XT cascade 4: 1 XT-PIC-XT 5: 18834539 XT-PIC-XT bttv0, Bt87x audio 7: 0 XT-PIC-XT parport0 8: 0 XT-PIC-XT rtc0 9: 1587422 XT-PIC-XT eth0 10: 12183885 XT-PIC-XT VIA686A 11: 77875422 XT-PIC-XT uhci_hcd:usb1, uhci_hcd:usb2 12: 7 XT-PIC-XT i8042 14: 17133058 XT-PIC-XT ide0 15: 0 XT-PIC-XT ide1 NMI: 0 Non-maskable interrupts LOC: 0 Local timer interrupts SPU: 0 Spurious interrupts RES: 0 Rescheduling interrupts CAL: 0 Function call interrupts TLB: 0 TLB shootdowns TRM: 0 Thermal event interrupts ERR: 0 MIS: 0 Rich Horner Engineering and Technology The Nielsen Company 813-366-5060 www.nielsen.com

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[alsa-devel] How to loopback two PCM interface?
by onlyfever 24 Jul '11

24 Jul '11

Hi, I configure McBSP3 and McBSP4 as PCM interface. I want to connect two PCM interface internally. The data captured from PCM1 can be directly sent to PCM2 playback interface. How can I achieve that? Please give me some advise. thank you!

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Re: [alsa-devel] sound: constify snd_ac97_bus_ops
by Takashi Iwai 24 Jul '11

24 Jul '11

At Sun, 24 Jul 2011 11:40:48 +0200, Lionel Debroux wrote: > > The fields of this struct type don't seem to be used as rvalues, besides > the initialization of struct instances. Let's make const the struct > instances. > > Maybe the Documentation/DocBook/writing-an-alsa-driver.tmpl, and > checkpatch.pl should be updated to reflect this change ? Yes, please either update or give an additional patch for document fixes. I'll apply these series not for 3.1 since this changes API. Will be applied the branches for 3.2. thanks, Takashi

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