June 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] No mixers on ATI RS780 Azalia
by Jan Engelhardt 07 Jun '10

07 Jun '10

Hi, I have here a Samsung 320MXn2 monitor/mainboard (typical x86_64 environment) where I cannot get sound output to the internal speakers working. To start, there seem to be two sound cards: lspci/lspci -n: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller 00:14.2 0403: 1002:4383 01:05.1 0403: 1002:960f /proc/asound/cards: 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xf0500000 irq 16 1 [HDMI]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xf0110000 irq 19 The soundcard responsible for the internal speaker is the 01:05.1/"HDMI" one. Under Windows XP, I see the following mixer elements for it: * Master Volume * Wave * SW Synth * CD Player Subsequently, there is sound (once I bump the volumes on these). However, in Linux up to including 2.6.34, I only see a single mixer element: * IEC958/S/PDIF mplayer etc. all successfully write to the device, but since there are not any further mixer controls, I can't unmute the essential channels. alsa-utils is version 1.0.21, from openSUSE 11.2 x86_64. Any help is appreciated. Thanks, Jan

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 07 Jun '10

07 Jun '10

The following changes since commit 1fab79b8a1bf0d46a7c7e6a1c06dbed1803f403c: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2010-05-31 18:13:20 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Eric Bénard (2): ASoC: imx-ssi.c: add new choices to platform configuration ASoC: imx: add eukrea-tlv320 Mark Brown (1): Merge branch 'for-2.6.35' into for-2.6.36 Ryan Mallon (1): ASoC: ep93xx i2s audio driver Sascha Hauer (2): ASoC: Add missing Kconfig entry for Phytec boards ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1 Wan ZongShun (10): ASoC: nuc900: patch for SUBSTREAM_TYPE', 'PCM_TX' and 'PCM_RX' removal ASoC: nuc900: fix a wait loop bug ASoC: nuc900: fix a typo and rename the header file ASoC: nuc900: patch for modifing the ac97 delays to minimum ASoC: s6000: use resource_size for {request/release}_mem_region and ioremap ASoC: use resource_size for au1x ASoC: atmel: patch for the unnecessary variable removal ASoC: s3c: patch for the unnecessary variable 'state' removal ASoC: atmel: trivial code cleanup ASoC: patch for the useless 'break' removal in kirkwood arch/arm/plat-mxc/include/mach/ssi.h | 3 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/atmel-pcm.c | 1 - sound/soc/atmel/atmel_ssc_dai.c | 1 - sound/soc/au1x/psc-ac97.c | 13 +- sound/soc/au1x/psc-i2s.c | 13 +- sound/soc/au1x/psc.h | 1 - sound/soc/ep93xx/Kconfig | 9 + sound/soc/ep93xx/Makefile | 8 + sound/soc/ep93xx/ep93xx-i2s.c | 487 ++++++++++++++++++++ sound/soc/ep93xx/ep93xx-i2s.h | 18 + sound/soc/ep93xx/ep93xx-pcm.c | 319 +++++++++++++ sound/soc/ep93xx/ep93xx-pcm.h | 22 + sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 2 + sound/soc/imx/eukrea-tlv320.c | 135 ++++++ sound/soc/imx/imx-ssi.c | 11 +- sound/soc/kirkwood/kirkwood-i2s.c | 1 - sound/soc/nuc900/nuc900-ac97.c | 32 +- .../soc/nuc900/{nuc900-auido.h => nuc900-audio.h} | 4 - sound/soc/nuc900/nuc900-pcm.c | 18 +- sound/soc/s3c24xx/s3c-ac97.c | 1 - sound/soc/s6000/s6000-i2s.c | 38 +- 24 files changed, 1080 insertions(+), 79 deletions(-) create mode 100644 sound/soc/ep93xx/Kconfig create mode 100644 sound/soc/ep93xx/Makefile create mode 100644 sound/soc/ep93xx/ep93xx-i2s.c create mode 100644 sound/soc/ep93xx/ep93xx-i2s.h create mode 100644 sound/soc/ep93xx/ep93xx-pcm.c create mode 100644 sound/soc/ep93xx/ep93xx-pcm.h create mode 100644 sound/soc/imx/eukrea-tlv320.c rename sound/soc/nuc900/{nuc900-auido.h => nuc900-audio.h} (95%)

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 07 Jun '10

07 Jun '10

Hi Takashi, Some mainly OMAP McBSP specific patches to reduce power from Peter. We have Tony's Ack on all the arch/arm/omap stuff. Thanks Liam The following changes since commit 1fab79b8a1bf0d46a7c7e6a1c06dbed1803f403c: Takashi Iwai (1): Merge branch 'for-2.6.36' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Peter Ujfalusi (6): OMAP: McBSP: Function to query the FIFO size OMAP3: McBSP: Change the way how the FIFO is handled OMAP3: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams ASoC: tlv320dac33: Add support for changing upper threshold arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 51 ++++++++++----- sound/soc/codecs/tlv320dac33.c | 57 +++++++++++++--- sound/soc/omap/omap-mcbsp.c | 112 +++++++++++++++++++++++------- 5 files changed, 176 insertions(+), 56 deletions(-)

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Re: [alsa-devel] [PATCH] ASoC/kirkwood: patch for the useless 'break' removal.
by Wan ZongShun 07 Jun '10

07 Jun '10

Hi Mark, How about this patch? My other patch seem to be applied, but this one still be hung. 在 2010年6月2日下午4:03，Wan ZongShun <mcuos.com(a)gmail.com> 写道： > This patch to remove the 'break;', when the 'switch' jumps to > the 'default' branch, the 'return -EINVAL' will be return with > a error number, so the 'break;' code never be run, it is unuseful > and should be removed here. > > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > --- > sound/soc/kirkwood/kirkwood-i2s.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c > index 0adc597..0fdc7db 100644 > --- a/sound/soc/kirkwood/kirkwood-i2s.c > +++ b/sound/soc/kirkwood/kirkwood-i2s.c > @@ -296,7 +296,6 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, > > default: > return -EINVAL; > - break; > } > > return 0; > -- > 1.6.3.3 > -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

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Re: [alsa-devel] [PATCH] Atmel/alsa: fix some code issues
by Wan ZongShun 07 Jun '10

07 Jun '10

在 2010年6月7日下午6:41，Nicolas Ferre <nicolas.ferre(a)atmel.com> 写道： > Le 28/05/2010 04:27, Wan ZongShun : >> Dear Nicolas, >> >> I am seeing the atmel codes as reference of my alsa driver programming. >> Finding a few code issues, I make a patch to fix, thanks! > > After a brief review, I come back to this > >> Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> >> >> --- >> sound/soc/atmel/atmel-pcm.c | 2 +- >> sound/soc/atmel/atmel_ssc_dai.c | 3 +-- >> 2 files changed, 2 insertions(+), 3 deletions(-) >> >> diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c >> index f6b3cc0..c69a8fb 100644 >> --- a/sound/soc/atmel/atmel-pcm.c >> +++ b/sound/soc/atmel/atmel-pcm.c >> @@ -478,7 +478,7 @@ struct snd_soc_platform atmel_soc_platform = { >> .pcm_free = atmel_pcm_free_dma_buffers, >> .suspend = atmel_pcm_suspend, >> .resume = atmel_pcm_resume, >> -}; >> +} > > Not ok for this. It is a structure declaration: we should keep the ";". > Okay, thanks! >> EXPORT_SYMBOL_GPL(atmel_soc_platform); >> >> static int __init atmel_pcm_modinit(void) >> diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c >> index 0b59806..4c44d13 100644 >> --- a/sound/soc/atmel/atmel_ssc_dai.c >> +++ b/sound/soc/atmel/atmel_ssc_dai.c >> @@ -549,7 +549,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream >> *substream, >> printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", >> ssc_p->daifmt); >> return -EINVAL; >> - break; > > Ok. > >> } >> pr_debug("atmel_ssc_hw_params: " >> "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", >> @@ -760,7 +759,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { >> .private_data = &ssc_info[2], >> }, >> #endif >> -}; >> +} > > Not ok for this either. ditto. > >> EXPORT_SYMBOL_GPL(atmel_ssc_dai); >> >> static int __init atmel_ssc_modinit(void) > > > -- > Nicolas Ferre > > -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

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Re: [alsa-devel] [PATCH v2 1/3]AsoC/nuc900: patch for SUBSTREAM_TYPE', 'PCM_TX' and 'PCM_RX' removal
by Liam Girdwood 07 Jun '10

07 Jun '10

On Wed, 2010-06-02 at 13:54 +0800, Wan ZongShun wrote: > This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition. > > There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX, > the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be > judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather > than 'if (PCM_TX == stype)', which makes the codes easy to read. > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> All Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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Re: [alsa-devel] [PATCH] sound/spi: patch for the unuseful variable removal
by Liam Girdwood 07 Jun '10

07 Jun '10

On Fri, 2010-06-04 at 16:30 +0800, Wan ZongShun wrote: > The '*bitclk' of structure 'snd_at73c213' seems no use > ,so I make a patch to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> > --- > sound/spi/at73c213.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c > index 4c7b051..1bc56b2 100644 > --- a/sound/spi/at73c213.c > +++ b/sound/spi/at73c213.c > @@ -69,7 +69,6 @@ struct snd_at73c213 { > int irq; > int period; > unsigned long bitrate; > - struct clk *bitclk; > struct ssc_device *ssc; > struct spi_device *spi; > u8 spi_wbuffer[2]; Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for changing upper threshold
by Peter Ujfalusi 07 Jun '10

07 Jun '10

Upper threshold is used in mode7 of DAC33. Instead of hard wired UTHR, add control to change the upper threshold value. Changing upper threshold is not allowed when the playback is already running, since wrongly timed change in the UTHR can cause problems with the codec. With this control the length of the burst in mode7 can be changed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 57 +++++++++++++++++++++++++++++++++------ 1 files changed, 48 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 65adc77..2fa946c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -120,6 +120,8 @@ struct tlv320dac33_priv { * samples */ unsigned int mode7_us_to_lthr; /* Time to reach lthr from uthr */ + unsigned int uthr; + enum dac33_state state; }; @@ -442,6 +444,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol, return ret; } +static int dac33_get_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = dac33->uthr; + + return 0; +} + +static int dac33_set_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (dac33->substream) + return -EBUSY; + + if (dac33->uthr == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) || + ucontrol->value.integer.value[0] > MODE7_UTHR) + ret = -EINVAL; + else + dac33->uthr = ucontrol->value.integer.value[0]; + + return ret; +} + static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -506,6 +541,8 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = { static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0, + dac33_get_uthr, dac33_set_uthr), SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum, dac33_get_fifo_mode, dac33_set_fifo_mode), }; @@ -985,7 +1022,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) * Configure the threshold levels, and leave 10 sample space * at the bottom, and also at the top of the FIFO */ - dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR)); + dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr)); dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR)); break; default: @@ -1052,8 +1089,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) break; case DAC33_FIFO_MODE7: dac33->mode7_us_to_lthr = - SAMPLES_TO_US(substream->runtime->rate, - MODE7_UTHR - MODE7_LTHR + 1); + SAMPLES_TO_US(substream->runtime->rate, + dac33->uthr - MODE7_LTHR + 1); dac33->t_stamp1 = 0; break; default: @@ -1104,7 +1141,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_soc_codec *codec = socdev->card->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; - unsigned int time_delta; + unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; @@ -1182,6 +1219,7 @@ static snd_pcm_sframes_t dac33_dai_delay( case DAC33_FIFO_MODE7: spin_lock(&dac33->lock); t0 = dac33->t_stamp1; + uthr = dac33->uthr; spin_unlock(&dac33->lock); t_now = ktime_to_us(ktime_get()); @@ -1194,7 +1232,7 @@ static snd_pcm_sframes_t dac33_dai_delay( * Either the timestamps are messed or equal. Report * maximum delay */ - delay = MODE7_UTHR; + delay = uthr; goto out; } @@ -1208,8 +1246,8 @@ static snd_pcm_sframes_t dac33_dai_delay( substream->runtime->rate, time_delta); - if (likely(MODE7_UTHR > samples_out)) - delay = MODE7_UTHR - samples_out; + if (likely(uthr > samples_out)) + delay = uthr - samples_out; else delay = 0; } else { @@ -1227,8 +1265,8 @@ static snd_pcm_sframes_t dac33_dai_delay( time_delta); delay = MODE7_LTHR + samples_in - samples_out; - if (unlikely(delay > MODE7_UTHR)) - delay = MODE7_UTHR; + if (unlikely(delay > uthr)) + delay = uthr; } break; default: @@ -1484,6 +1522,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; dac33->nsample_max = NSAMPLE_MAX; + dac33->uthr = MODE7_UTHR; /* Disable FIFO use by default */ dac33->fifo_mode = DAC33_FIFO_BYPASS; -- 1.7.1

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[alsa-devel] 2.6.34-rc: sound hungs (ice1724)
by Alexander Beregalov 06 Jun '10

06 Jun '10

Hi Takashi Could you please help me how to debug an issue. Sometimes soung hungs, it happens few days (up to a week) after boot. I have no idea how to reproduce it or how to debug it. That's why it is very hard to bisect, I have not tried, the better way should exist. The problem appeared somewhere after 2.6.33. Card is M-Audio Revo 5.1 (1412:1724). After it hungs ALSA continues to play last sound frame forever. When I try to play something else playing frame changes. Only reboot heals it. rmmod hungs at INFO: task rmmod:30356 blocked for more than 120 seconds. "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. rmmod D db281e7c 0 30356 2345 0x00000004 db281e8c 00000086 00000002 db281e7c cc2d7b90 00000000 c14e8b80 09c3638a c10252a0 00000001 c14e2000 c14e2000 c14e8b80 c14e8b80 cc2d7a00 db281e8c 00000246 db281e8c c1043d23 00000001 f6923000 f69231b4 db281e9c db281ebc Call Trace: [<c10252a0>] ? __wake_up+0x40/0x50 [<c1043d23>] ? prepare_to_wait+0x43/0x70 [<f8ce328d>] snd_card_free+0x5d/0x90 [snd] [<c1043af0>] ? autoremove_wake_function+0x0/0x40 [<f9ebcaa6>] snd_vt1724_remove+0x13/0x25 [snd_ice1724] [<c11cddb9>] pci_device_remove+0x19/0x40 [<c1239f3e>] __device_release_driver+0x3e/0x90 [<c123a017>] driver_detach+0x87/0x90 [<c1239323>] bus_remove_driver+0x63/0xa0 [<c123a526>] driver_unregister+0x46/0x80 [<c10d616d>] ? sysfs_remove_file+0xd/0x10 [<c11cdfed>] pci_unregister_driver+0x2d/0x80 [<f9ebca91>] alsa_card_ice1724_exit+0xd/0xf [snd_ice1724] [<c1057bee>] sys_delete_module+0x15e/0x210 [<c107f91f>] ? remove_vma+0x3f/0x60 [<c1080902>] ? do_munmap+0x202/0x2d0 [<c1002b50>] sysenter_do_call+0x12/0x26 CONFIG_SOUND=y CONFIG_SND=m CONFIG_SND_TIMER=m CONFIG_SND_PCM=m CONFIG_SND_RAWMIDI=m CONFIG_SND_VMASTER=y CONFIG_SND_DMA_SGBUF=y CONFIG_SND_AC97_CODEC=m CONFIG_SND_PCI=y CONFIG_SND_ICE1724=m

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[alsa-devel] multi plugin and start_threshold = -1
by Raymond Yau 06 Jun '10

06 Jun '10

I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. I: alsa-sink.c: Starting playback. I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed (-77) I: (alsa-lib)pcm_hw.c: PCM state = RUNNING any limitation to use -1 as start_thershold ? why SNDRV_PCM_IOCTL_START failed (-77) only occur with PA server when using "multi" plugin of surround40 of ca0106 ? other application (e.g. aplay , speaker-test and mplayer seem work fine https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5018

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