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[alsa-devel] usb-audio code refactoring, more v2 support [resend]
by Daniel Mack 05 Mar '10

05 Mar '10

This is a rebased version of the usb-audio cleanup patch set. On top of what's already been posted, there is one more fix to support sample rate configuration, and one bugfix. Commit 864c11080 was also ported over. And Clemens' latest patches are also included. With these patches applied, I can stream audio on a class v2 device. At least for awhile, then I get CRC errors in DATA0 packets. But that's a different issue. What's not yet done is all the feature unit parsing, mixers etc. Daniel [PATCH 1/8] ALSA: usb-audio: move ua101 driver [PATCH 2/8] ALSA: usb-audio: header file cleanups [PATCH 3/8] ALSA: usb-audio: refactor code [PATCH 4/8] ALSA: usb-audio: rename substream format field to altset_idx [PATCH 5/8] ALSA: usb-audio: use a format bitmask per alternate setting [PATCH 6/8] ALSA: usb-audio: support multiple formats with audio class v2 devices [PATCH 7/8] ALSA: usb-audio: add support for samplerate setting on v2 devices [PATCH 8/8] usb/audio.h: Fix field order

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[alsa-devel] [PATCH 4/8] ALSA: usb-audio: rename substream format field to altset_idx
by Daniel Mack 05 Mar '10

05 Mar '10

From: Clemens Ladisch <clemens(a)ladisch.de> The snd_usb_substream::format field actually contains the index of the current alternate setting, so rename it to altset_idx to avoid confusion. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/card.h | 2 +- sound/usb/pcm.c | 8 ++++---- sound/usb/proc.c | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 71f03c1..856d71b 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -54,7 +54,7 @@ struct snd_usb_substream { struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ - unsigned int format; /* USB data format */ + unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int datapipe; /* the data i/o pipe */ unsigned int syncpipe; /* 1 - async out or adaptive in */ unsigned int datainterval; /* log_2 of data packet interval */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 87863cc..c3d5a97 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -202,11 +202,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) return -EIO; } subs->interface = -1; - subs->format = 0; + subs->altset_idx = 0; } /* set interface */ - if (subs->interface != fmt->iface || subs->format != fmt->altset_idx) { + if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", dev->devnum, fmt->iface, fmt->altsetting); @@ -214,7 +214,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); subs->interface = fmt->iface; - subs->format = fmt->altset_idx; + subs->altset_idx = fmt->altset_idx; } /* create a data pipe */ @@ -771,7 +771,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; subs->interface = -1; - subs->format = 0; + subs->altset_idx = 0; runtime->hw = snd_usb_hardware; runtime->private_data = subs; subs->pcm_substream = substream; diff --git a/sound/usb/proc.c b/sound/usb/proc.c index be3065e..78fc3ba 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -116,7 +116,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn unsigned int i; snd_iprintf(buffer, " Status: Running\n"); snd_iprintf(buffer, " Interface = %d\n", subs->interface); - snd_iprintf(buffer, " Altset = %d\n", subs->format); + snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); for (i = 0; i < subs->nurbs; i++) snd_iprintf(buffer, "%d ", subs->dataurb[i].packets); -- 1.6.6.2

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[alsa-devel] [PATCH] fix jazz16 compile (udelay)
by Meelis Roos 05 Mar '10

05 Mar '10

While trying to compile jazz16 isa sound driver on alpha (2.6.33+git), I found a compile failure in jazz16.c (udelay is unknown). Fix it by including delay.h. Signed-foo-by: Meelis Roos <mroos(a)linux.ee> diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 8d21a3f..8ccbcdd 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -14,6 +14,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/io.h> +#include <linux/delay.h> #include <asm/dma.h> #include <linux/isa.h> #include <sound/core.h> -- Meelis Roos (mroos(a)linux.ee)

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Re: [alsa-devel] [RFC 0/4] FIFO caused playback delay (latency) handling in soc
by Raymond Yau 05 Mar '10

05 Mar '10

>> Do soc support snd_pcm_rewind() and snd_pcm_forward ? >>No driver demand yet, though it'd be trivially implemented if anyone >>wanted to do it. if size of fifo is around 1x ~ 3x fames How accucate do those media player need to allow A/V lip sync , Video is about 24 ~ 30 frames per seconds and audio is 48000Hz ?

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[alsa-devel] snd-powermac @ G4 DA (PowerMac3, 4) - headphone detection doesn't work
by Douglas Mencken 04 Mar '10

04 Mar '10

I can hear sound with internal speakers only if I plug headphones jack, without plugged jack, there's no sound at all; to listen thru headphones, I need to half-insert that jack and this leads to distortions; with h/phones jack plugged fully, no sound in phones, but sound in internal speaker. With the same half-plug, in OS X I head sound both via h/phones and internal speaker. Driver: snd-powermac. Audio card: Tumbler @ i2ca @ KeyLargo i2c (I can provide the full dump of OpenFirmware device tree). As <gnubien> at #alsa @ irc.freenode.net suggested, I ran `alsamixer -c 0` and MM-ed 'Auto Mute'. After that, I can fully plug h/phones jack and enjoy full sounds w/o distortions but still can't get any sound when it's unplugged. The result of running ./alsa-info.sh: Your ALSA information is located at http://www.alsa-project.org/db/?f=daac9e06cbe772222b90d7fba14c1a49a65b980c

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[alsa-devel] [RFC] Cleanup/refactor the usbaudio driver
by Daniel Mack 04 Mar '10

04 Mar '10

I've spent some time cleaning up the usbaudio driver. I felt that the existing implementation was quite difficult to follow because it contained code for all sorts of things mixed up in one big file. So I split the driver into pieces and tried to be as less intrusive as I can, meaning that I didn't touch much real code but rather copied the functions as they were and factored them out. Of course, non-static functions were properly prefixed to avoid name space pollution. I also moved almost all quirks to a seperate file and introduced two more generic hooks to handle them, so the excecption add less to the actual code of the driver. Unfortunately it wasn't easily possible to put that big change into smaller pieces, and so 3/3 became a huge chunk. After all, I feel that it is easier to read now, but I don't know whether you agree. So take this as RFC for now :) Daniel

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for Dell Latitude 131L
by Daniel Chen 04 Mar '10

04 Mar '10

Hi, Please apply the following patch against sound-2.6 master HEAD (applies with tweaking against 2.6.32.x) to work around hardware issues on Dell Latitude 131L. Best, -Dan

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[alsa-devel] [PATCH] ASoC: Add delay information for Samsung IISv2 DAIs
by Mark Brown 04 Mar '10

04 Mar '10

Report the current FIFO depth when delay is queried. The FIFO is only 16 frames deep so the latency will be at most a couple of miliseconds (and we tend to end up reporting zero most of the time) but it may help some applications. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index e994d83..04a00fa 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -550,6 +550,21 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, return 0; } +static long s3c2412_i2s_delay(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 reg = readl(i2s->regs + S3C2412_IISFIC); + int delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = S3C2412_IISFIC_TXCOUNT(reg); + else + delay = S3C2412_IISFIC_RXCOUNT(reg); + + return delay; +} + /* default table of all avaialable root fs divisors */ static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; @@ -736,6 +751,10 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) ops->set_fmt = s3c2412_i2s_set_fmt; ops->set_clkdiv = s3c2412_i2s_set_clkdiv; + /* Allow overriding by (for example) IISv4 */ + if (!ops->delay) + ops->delay = s3c2412_i2s_delay, + dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; -- 1.7.0

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[alsa-devel] [RFC 0/4] FIFO caused playback delay (latency) handling in soc
by Peter Ujfalusi 04 Mar '10

04 Mar '10

Hello, There has been discussion in alsa-devel in this subject several times, but no actual patches has been sent. Based on the available information the latency caused by the HW buffer on some systems can be handled by updating the runtime->delay. It has been discussed, that the runtime->delay can also be updated dynamically to show more accurate delay. To further complicate things, in ASoC we could have more buffer in the chain. To handle this we need soc level support. This RFC series tries to do that in soc by: - introducing a pcm_pointer wrapper - in this wrapper we call the original pcm_pointer functions to get the DMA pointer - introducing a new interface in dai_ops to ask the delay from the dais - adding the cpu_dai and codec_dai returned delay to form the actual delay - update the runtime->delay with this value. With this approach none of the existing drivers need change, but they can add support for specifying the FIFO caused delay. In this series on top of the core changes the omap(3) code is updated to take this delay reporting into use. I have not added the support to the tlv320dac33 codec driver, since it needs a bit more work, but along the same line it can be done, and if the tlv320dac33 is hooked to omap McBSP than applications can know the whole delay/latency on that path. Since this is for RFC, I have based it on sound-2.6 topc/asoc branch, which does not have the McBSP regcache nor the sidetone patches, but I would like to get feedback on this method first than create a proper series. I have left out linux-omap from this RFC round, since the problem is ALSA related. Thank you, Péter Ujfalusi --- Peter Ujfalusi (4): ASoC: core: soc level wrapper for pcm_pointer callback ASoC: core: Add delay operation to snd_soc_dai_ops OMAP3: McBSP: Add interface for transmit FIFO state query ASoC: OMAP3: Report delay on playback caused by the internal FIFO arch/arm/plat-omap/include/plat/mcbsp.h | 4 +++ arch/arm/plat-omap/mcbsp.c | 27 ++++++++++++++++++++++++ include/sound/soc-dai.h | 6 +++++ sound/soc/omap/omap-mcbsp.c | 26 +++++++++++++++++++++++ sound/soc/soc-core.c | 34 ++++++++++++++++++++++++++++++- 5 files changed, 96 insertions(+), 1 deletions(-)

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 03 Mar '10

03 Mar '10

The following changes since commit a0b62329bb290c10d7278809af910ed115768991: Takashi Iwai (1): Merge branch 'for-2.6.34' of git://opensource.wolfsonmicro.com/linux-2.6-asoc into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (1): ASoC: fix ak4104 register array access Jassi Brar (1): ASoC: soc_pcm_open: Add missing bailout tag sound/soc/codecs/ak4104.c | 6 ++---- sound/soc/soc-core.c | 14 +++++++++----- 2 files changed, 11 insertions(+), 9 deletions(-)

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