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[alsa-devel] ASoC update for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 87831cb660954356d68cebdb1406f3be09e784e9: Mark Brown (1): ASoC: Fix WM835x Out4 capture enumeration are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-)

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[alsa-devel] [PATCH] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ 1 files changed, 96 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); -- 1.6.4.2

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[alsa-devel] Update to problem with Creative X-fi mixer problems
by Susan Cragin 08 Sep '09

08 Sep '09

I saw that someone was going to try to get all the mixer controls to work properly. I updated and opened alsamixer this morning, and found that both the incoming/recording controls were there! (I should have quit while I was ahead.) Then I re-booted and recompiled wine and opened winecfg and found that oss could find the proper mixer but alsa did not. (The oss mixer is listed as Creative CA0110-IBG) ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm_dsnoop.c:593:(snd_pcm_dsnoop_open) unable to open slave fixme:mixer:ALSA_MixerInit No master control found on HD-Audio Generic, disabling mixer susan@ubuntu:~$ alsamixer No mixer elems found

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Re: [alsa-devel] snd_dummy on Centos - Redhat
by Takashi Iwai 08 Sep '09

08 Sep '09

At Fri, 4 Sep 2009 00:43:09 +0200, Giovanni Maruzzelli wrote: > > On Fri, Sep 4, 2009 at 12:03 AM, Giovanni > Maruzzelli<gmaruzz.lists(a)gmail.com> wrote: > > I'll test it and report back here. > > Still we have big problems.... OK, then upgrade your alsa driver stack to 1.0.21. There have been some changes in PCM core. Then, get dummy.c again from the latest alsa-driver-unstable tarball. It already contains my latest fix. If 1.0.21+dummy.c doesn't work, apply the patch below to get position logs. If this still causes the invalid backward jump to position 0, you should check xrun debug, e.g. # echo 9 > /proc/asound/card0/pcm0p/xrun_debug (assuming snd-dummy is card0). Takashi --- diff --git a/dummy.c b/dummy.c index e8e29bf..f7c21d5 100644 --- a/dummy.c +++ b/dummy.c @@ -239,6 +239,10 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) dpcm->frac_period_rest += dpcm->frac_period_size; } dpcm->frac_period_rest -= delta; + printk(KERN_DEBUG "xxx delta=%ld, pos=%d(%d), rest=%d(%d)\n", + delta / dpcm->rate, + dpcm->frac_pos, dpcm->frac_pos / HZ, + dpcm->frac_period_rest, dpcm->frac_period_rest / HZ); } static int dummy_systimer_start(struct snd_pcm_substream *substream) @@ -248,6 +252,10 @@ static int dummy_systimer_start(struct snd_pcm_substream *substream) dpcm->base_time = jiffies; dummy_systimer_rearm(dpcm); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** START: buffer=%d, period=%d, rate=%d\n", + (int)substream->runtime->buffer_size, + (int)substream->runtime->period_size, + substream->runtime->rate); return 0; } @@ -257,6 +265,7 @@ static int dummy_systimer_stop(struct snd_pcm_substream *substream) spin_lock(&dpcm->lock); del_timer(&dpcm->timer); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** STOP\n"); return 0; }

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[alsa-devel] Audio cutoff when playing using dmix
by ruchi sirauthiya 08 Sep '09

08 Sep '09

Hi, I am facing audio clipping issues when dmix is used for playback. For the test we used a full scale sine wave tone generated using software and encoded as "Signed 16 bit Little Endian, Rate 44100 Hz, Mono". The audio is getting cut out at a 2-3 second interval during playback. This happens only for a fraction of second but it is consistent. This does not happen when audio is played back on the device directly but only when going through dmix. We have observed that there were differences in audio cutting off when the sampling rate was set to 8000Hz and 32000 Hz in asound.conf. The contents of asound.conf is pasted below for your reference: pcm.dmixer { type dmix ipc_key 2048 ipc_key_add_uid true slave { pcm hw:0 rate 8000 channels 2 } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "hw:0,0" } pcm.rateConvert { type plug slave.pcm "asymed" } The configuration given above works fine with dmix in 2.6.29 kernel, I am using kernel 2.6.22.18. I have compared the configuration differences (buffers and periods) with and without using dmix. I tried applying some parameters (like buffer_size, period_time and period_size) which was in use when audio was played directly to the hardware device by aplay using the following command: aplay -D rateConvert --period-time=92879 --buffer-size=65536 --period-size=4096 /mnt/test.wav -vvv -d 5 There was come improvement in the audio cutting off issue but it was not eliminated completely. Given below are the logs for aplay (-vvv) with and without using dmix (with audio cutoff issue on kernel 2.6.22): Without using dmix: [root@OMAP3 tests]# aplay /mnt/test.wav -vvv -d 5 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 65536 period_size : 4096 period_time : 92879 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 65536 stop_threshold : 65536 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% ...Same text repeats 50 times... Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% Using dmix: [root@OMAP3 tests]# aplay -D rateConvert /mnt/test.wav -vvv -d 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Rate conversion PCM (8000, sformat=S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 5512 xfer_align : 5512 start_threshold : 22048 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 16000 period_size : 1000 period_time : 125000 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 1 stop_threshold : 2097152000 silence_threshold: 0 silence_size : 2097152000 boundary : 2097152000 Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% ...Same text repeats 36 times... Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00005f1a ############### 74% Please help me fix this issue. Does this occur due to non-optimal settings of the parameters (buffers, periods) in the configuration or is it some other issue with the ALSA library or the kernel? Please ask if any more setup/configuration settings are required for reference. Thanks and Regards, Ruchi Sirauthiya

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 367da1527aff2a4f789d03082f9fb191cdf332c4: Robert Schwebel (1): ASoC: fix pxa2xx-ac97.c breakage are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Mark Brown (2): ASoC: Remove unuused hw_read_t ASoC: Fix WM835x Out4 capture enumeration include/sound/soc.h | 1 - sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 1 insertions(+), 2 deletions(-)

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Re: [alsa-devel] Patch for au88x0 driver. Fix audacious issue and add auto reset in case of hardware ADB ctrl failure.
by Raymond Yau 08 Sep '09

08 Sep '09

2009/9/7 Manuel Jander <manuel.jander(a)gmail.com> > Hi Raymond, > > On Mon, Sep 7, 2009 at 3:47 AM, Raymond Yau<superquad.vortex2(a)gmail.com> > wrote: > > On Thursday 03 September 2009 21:10, Manuel Jander wrote: > >> does anybody still have a AU88x0 card ? I know they are quite ancient > >> now, but if anyone could give the attached patch a try ? Would be > >> nice. > >> I made the patch against linux-2.6.30.4 > >> http://micro.homelinux.net/~mjander/au88x0.diff.gz<http://micro.homelinux.net/%7Emjander/au88x0.diff.gz>< > http://micro.homelinux.net/%7Emjander/au88x0.diff.gz> > >> > >> Best Regards, > >> Manuel > > > > Which version of audaicous (1.5.x or 2.1) are you testing ? > > (alsa-gapless or alsa-ng plugin) > > > > au88x0 most likely cannot support 1 sample per period because of the > > FIFO as described in patent 6167465. > > Well, that is clear to me now. Using a DMA transfer that is smaller > than one audio sample is not really a good idea. > I would be better off using PIO instead of DMA. > > > if you look at the source code of alsa-ng and alsa-gapless plugin > > both of them use the own way to calculate the buffer size instead of > > using snd_pcm_hw_params_get_buffer_size() > > > > > > > http://hg.atheme.org/audacious-plugins/file/2c0a9bba77d7/src/alsa-ng/alsa-c… > > > > CHECK_FAIL (snd_pcm_hw_params_set_access (pcm_handle, hwparams, > > SND_PCM_ACCESS_RW_INTERLEAVED), "snd_pcm_hw_params_set_access"); > > CHECK_FAIL (snd_pcm_hw_params_set_format (pcm_handle, hwparams, afmt), > > "snd_pcm_hw_params_set_format"); > > CHECK_FAIL (snd_pcm_hw_params_set_channels (pcm_handle, hwparams, nch), > > "snd_pcm_hw_params_set_channels"); > > CHECK_FAIL (snd_pcm_hw_params_set_rate (pcm_handle, hwparams, rate, 0), > > "snd_pcm_hw_params_set_rate"); > > CHECK_FAIL (snd_pcm_hw_params (pcm_handle, hwparams), > "snd_pcm_hw_params"); > > bitwidth = snd_pcm_format_physical_width(afmt); > > bps = (rate * bitwidth * nch) >> 3; > > buf_size = MAX(aud_cfg->output_buffer_size, 100); > > ringbuf_size = buf_size * bps / 1000; > > I increased the minimum period bytes value to 256 in that patch. > Audacious version 2 (Debian SID) > works for me. > well, audacious.i386 1.5.1-9.fc10 cannot play any music (alsa plugin) on my au8830 with your patch, it only play music with (oss plugin) Use xmms instead of audacious since they have similar UI 256 seem much bigger than the figures mentioned in the patent , may be it is the practical limit on your VIA ITX In one embodiment of the present invention, the FIFO buffer is 32 samples or words in depth, which would withstand a maximum bus latency of about 4 microseconds. By way of example, if 48 DMA channels trigger simultaneously, with a trigger level set at half the buffer size (16 samples), each DMA channel can wait 16 samples (20 microseconds per sample), or approximately 320 microseconds before running out of data. By way of example, CPU 210 may update-sub-buffer tables 380(0)-380(x) routinely every 10 milliseconds. The periodic checks may be initiated by audio processing peripheral 230 by way of interrupts, or may be scheduled independently by the driver > > How do you get the "ADB Ctrl failure" error ? > > I do not know how to reproduce that. Sometimes it happens, and its not > possible > to recover without a hardware reset (unload and load module). Somehow > it semes to > happen when my computer gets really hot (watching an MPEG-4 video, or > compiling the kernel...). > Its a old VIA ITX board, with a 1.3GHz processor (on demand cpufreq). > > The patch tries to reset the audio card if all PCM streams are > released once such a > condition was detected. This is the situation were the windows driver > used to crash the > whole machine. > it work quite well on my two motherbroad using intel chipset > > There are two main problem of the au88x0 driver > > 1) alsa is unable to calculate the correct value for the three > > functions snd_pcm_hw_params_set_buffer_time_near() , > > snd_pcm_hw_params_set_period_time_near() and snd_pcm_set_params() > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1724 > > > > Test case : > > > > alsa-lib/test/pcm_min.c ( change default to hw:0,0 ) > > > > Real world case: > > > > gstreamer and winealsa.drv > > I will try that. I had no idea about that bugs. > A simple workaround is add another constraint snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096 ); however winealsa.drv will be using a larger period bytes than wineoss.drv > > > 2) unable to recover when the number of available src is not enough > > (FATAL: ResManager: resource type 1 exhaust) > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2228 > > > > test case : > > > > running application to use 15 src ( 7 stereo and one mono stream ) and > > open another stereo stream > > Hmm, the good ol' resource manager :) OK, I will try to find a > solution. I have not much > time these days, but maybe it rains again, and I get bored, who knows :D > > Best Regards, > Manuel > simple fix may be adding counts of the resource and check that all resources are enough before allocating them

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[alsa-devel] Duplicate configuration nodes
by Jochem Wichers Hoeth 07 Sep '09

07 Sep '09

Hi, Since I upgraded from alsa-lib 1.0.20 to 1.0.21 my configuration is not parsed correctly anymore: $ aplay test.wav ALSA lib conf.c:4639:(snd_config_expand) Expand error (walk): File exists Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono As a result of this error, ALSA seems to ignore the second half of HDA-Intel.conf, including the iec958 definition. I traced the "File exists" error back to snd_config_add() returning -EEXIST, while evaluating the "strings" node at line 36 of cards/HDA-Intel.conf. I verified that at that point the src node indeed has four children: two duplicates of each child node of the original strings array. I'm not exactly sure what's happening here, but I do see that the HDA-Intel.conf file (and all of its included files) are actually read *twice* by aplay. With alsa-lib 1.0.20 aplay only reads these files once. The problem seems to be related to commit 96da0c842d14b40ce8e37726b259229634b3aa21 ("Fix driver conf parsing in snd_config_hook_load_for_all_cards()"), which subtly changes the logic when the same driver is encountered for the second time. I have two cards using the HDA-Intel config (one analog and one digital). For the first card, the config is loaded without issues. But for the second card, snd_config_search(root, fdriver, &n) succeeds but snd_config_get_string(n, &driver) fails. In 1.0.20 this results in a "goto __err", skipping the actual config load. But in 1.0.21 this results in the else-branch being chosen (driver = fdriver) and next snd_config_hook_load being called. I know too little of the entire config loading process to be sure what exactly is going wrong here, but a assume that the second call to snd_config_hook_load for the same configaration file somehow eventually results in the duplicate node error in snd_config_add(). I've verified that reverting 96da0c842d14b40ce8e37726b259229634b3aa21 fixes the problem, or at least the symptoms. :-) Please let me know if you need extra information, of if you'd like me to do some more testing. Cheers, Jochem Wichers Hoeth

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[alsa-devel] [PATCH] Minor WM8580 enhancements.
by jassi brar 07 Sep '09

07 Sep '09

Add more options to select clock source for various blocks in WM8580. Also, explicitly set defaults for the sake of consistency. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/codecs/wm8580.c | 133 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8580.h | 25 +++++---- 2 files changed, 144 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..276bc8e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -95,6 +95,7 @@ /* PLLB4 (register 7h) */ #define WM8580_PLLB4_MCLKOUTSRC_MASK 0x60 +#define WM8580_PLLB4_MCLKOUTSRC_MCLK 0x00 #define WM8580_PLLB4_MCLKOUTSRC_PLLA 0x20 #define WM8580_PLLB4_MCLKOUTSRC_PLLB 0x40 #define WM8580_PLLB4_MCLKOUTSRC_OSC 0x60 @@ -106,9 +107,16 @@ /* CLKSEL (register 8h) */ #define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_MCLK 0x00 #define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 #define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK 0x00 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLA 0x04 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLB 0x08 +#define WM8580_CLKSEL_ADC_CLKSEL_MCLK 0x0c + /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 #define WM8580_AIF_RATE_128 0x0 @@ -121,15 +129,16 @@ #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 -#define WM8580_AIF_BCLKSEL_128 0x08 -#define WM8580_AIF_BCLKSEL_256 0x10 +#define WM8580_AIF_BCLKSEL_32 0x08 +#define WM8580_AIF_BCLKSEL_16 0x10 #define WM8580_AIF_BCLKSEL_SYSCLK 0x18 #define WM8580_AIF_MS 0x20 #define WM8580_AIF_CLKSRC_MASK 0xc0 +#define WM8580_AIF_CLKSRC_ADCMCLK 0x00 #define WM8580_AIF_CLKSRC_PLLA 0x40 -#define WM8580_AIF_CLKSRC_PLLB 0x40 +#define WM8580_AIF_CLKSRC_PLLB 0x80 #define WM8580_AIF_CLKSRC_MCLK 0xc0 /* AIF control 2 (registers ch-eh) */ @@ -604,12 +613,13 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: - /* Input */ + reg |= WM8580_PLLB4_MCLKOUTSRC_MCLK; break; case WM8580_CLKSRC_PLLA: reg |= WM8580_PLLB4_MCLKOUTSRC_PLLA; break; + case WM8580_CLKSRC_PLLB: reg |= WM8580_PLLB4_MCLKOUTSRC_PLLB; break; @@ -630,6 +640,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_DAC_CLKSEL_MCLK; break; case WM8580_CLKSRC_PLLA: @@ -646,6 +657,59 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_CLKSEL, reg); break; + case WM8580_ADC_CLKSEL: + reg = snd_soc_read(codec, WM8580_CLKSEL); + reg &= ~WM8580_CLKSEL_ADC_CLKSEL_MASK; + + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_MCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLB; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_CLKSEL, reg); + break; + + case WM8580_PAIF_CLKSEL: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_CLKSRC_MASK; + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_AIF_CLKSRC_ADCMCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_AIF_CLKSRC_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_AIF_CLKSRC_PLLB; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_AIF_CLKSRC_MCLK; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -672,6 +736,67 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; + case WM8580_MCLKRATIO: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_RATE_MASK; + switch(div) { + case 128: + reg |= WM8580_AIF_RATE_128; + break; + + case 192: + reg |= WM8580_AIF_RATE_192; + break; + + case 256: + reg |= WM8580_AIF_RATE_256; + break; + + case 384: + reg |= WM8580_AIF_RATE_384; + break; + + case 512: + reg |= WM8580_AIF_RATE_512; + break; + + case 768: + reg |= WM8580_AIF_RATE_768; + break; + + case 1152: + reg |= WM8580_AIF_RATE_1152; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + + case WM8580_BCLKRATIO: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_BCLKSEL_MASK; + switch(div) { + case 64: + reg |= WM8580_AIF_BCLKSEL_64; + break; + + case 32: + reg |= WM8580_AIF_BCLKSEL_32; + break; + + case 16: + reg |= WM8580_AIF_BCLKSEL_16; + break; + + default: + reg |= WM8580_AIF_BCLKSEL_SYSCLK; + break; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + default: return -EINVAL; } diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..0ecdbfa 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -18,16 +18,21 @@ #define WM8580_PLLA 1 #define WM8580_PLLB 2 -#define WM8580_MCLK 1 -#define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 - -#define WM8580_CLKSRC_MCLK 1 -#define WM8580_CLKSRC_PLLA 2 -#define WM8580_CLKSRC_PLLB 3 -#define WM8580_CLKSRC_OSC 4 -#define WM8580_CLKSRC_NONE 5 - +#define WM8580_MCLK 1 +#define WM8580_DAC_CLKSEL 2 +#define WM8580_ADC_CLKSEL 3 +#define WM8580_PAIF_CLKSEL 4 +#define WM8580_CLKOUTSRC 5 +#define WM8580_MCLKRATIO 6 +#define WM8580_BCLKRATIO 7 + +#define WM8580_CLKSRC_ADCMCLK 0 +#define WM8580_CLKSRC_MCLK 1 +#define WM8580_CLKSRC_PLLA 2 +#define WM8580_CLKSRC_PLLB 3 +#define WM8580_CLKSRC_OSC 4 +#define WM8580_CLKSRC_NONE 5 + #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 -- 1.6.2.5

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[alsa-devel] [PATCH] ASoC: Factor out I2C 8 bit address 8 bit data I/O
by Joonyoung Shim 07 Sep '09

07 Sep '09

This patch is for the AK4671 codec driver using this format. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/soc-cache.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c8ceddc..404231e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -77,6 +77,35 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data, #define snd_soc_7_9_spi_write NULL #endif +static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -151,6 +180,7 @@ static struct { unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read }, + { 8, 8, snd_soc_8_8_write, NULL, snd_soc_8_8_read, NULL }, { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read, snd_soc_8_16_read_i2c }, }; -- 1.6.0.4

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