September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] Add quirks for Vaio VGN
by Pascal Terjan 28 Sep '09

28 Sep '09

Here are some quirks for 3 Vaio VGN model reported on launchpad to need model=vaio Pascal Terjan (3): [ALSA] Add quirk for Sony Vaio VGN-SZ650N [ALSA] Add quirk for Sony Vaio VGN-AR51M [ALSA] Add quirk for Sony Vaio VGN-FZ15G sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-)

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[alsa-devel] ctl_elems hooks with dmix plugin
by George Kuriakose 28 Sep '09

28 Sep '09

Hi I was trying to use ctl_elems hooks with dmix plugin but figured that this combination doesn't work. Is there any workaround for this. I am trying the following. There are two softvol devices s0 and s1 going through dmix plugins. I wan't to mute s0 when any app opens s1 and unmute s0 when s1 is closed. I am using the below asoundrc file which results in "No card for this PCM" . Any workarounds or any minor change in libasound which will make this work? Thanks George asoundrc: pcm.softvol0 { type softvol slave { pcm "dmix:1" } control { name "SoftControl0" card 1 } } pcm.softvol1 { type softvol slave { pcm "dmix:1" } control { name "SoftControl1" card 1 } } pcm.s0 { type plug slave.pcm "softvol0" } pcm.s1 { type plug slave.pcm "softvol1" } pcm.s1hook { type hooks # PCM with hooks slave.pcm "s1" hooks.0 { type ctl_elems hook_args [ { name "SoftControl0" preserve true lock true optional true value [ 0 ] } { name "SoftControl1" preserve true lock true optional true value [ 100 ] } ] } }

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[alsa-devel] [PATCH] sscape: add supoort for SPEA Media FX/Reveal SC-600
by Krzysztof Helt 28 Sep '09

28 Sep '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move code from the OSS sscape driver in order to support old Soundscape OEM models. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Only PCM works. sound/isa/Kconfig | 6 ++- sound/isa/sscape.c | 116 ++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 86 insertions(+), 36 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 51a7e37..b90fc16 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -377,10 +377,12 @@ config SND_SSCAPE select SND_WSS_LIB help Say Y here to include support for Ensoniq SoundScape - soundcards. + and Ensoniq OEM soundcards. The PCM audio is supported on SoundScape Classic, Elite, PnP - and VIVO cards. The MIDI support is very experimental. + and VIVO cards. The supported OEM cards are SPEA Media FX and + Reveal SC-600. + The MIDI support is very experimental. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 6618712..b11c35f 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -127,7 +127,8 @@ enum GA_REG { enum card_type { - SSCAPE, + MEDIA_FX, /* Sequoia S-1000 */ + SSCAPE, /* Sequoia S-2000 */ SSCAPE_PNP, SSCAPE_VIVO, }; @@ -784,20 +785,25 @@ static struct snd_kcontrol_new midi_mixer_ctl = { * These IRQs are encoded as bit patterns so that they can be * written to the control registers. */ -static unsigned __devinit get_irq_config(int irq) +static unsigned __devinit get_irq_config(int sscape_type, int irq) { static const int valid_irq[] = { 9, 5, 7, 10 }; + static const int old_irq[] = { 9, 7, 5, 15 }; unsigned cfg; - for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) { - if (irq == valid_irq[cfg]) - return cfg; - } /* for */ + if (sscape_type == MEDIA_FX) { + for (cfg = 0; cfg < ARRAY_SIZE(old_irq); ++cfg) + if (irq == old_irq[cfg]) + return cfg; + } else { + for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) + if (irq == valid_irq[cfg]) + return cfg; + } return INVALID_IRQ; } - /* * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. @@ -842,11 +848,39 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e) goto _done; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + if (s->ic_type == IC_OPUS) + activate_ad1845_unsafe(s->io_base); if (s->type == SSCAPE_VIVO) wss_io += 4; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + + /* wait for WSS codec */ + for (d = 0; d < 500; d++) { + if ((inb(wss_io) & 0x80) == 0) + break; + spin_unlock_irqrestore(&s->lock, flags); + msleep(1); + spin_lock_irqsave(&s->lock, flags); + } + snd_printd(KERN_INFO "init delay = %d ms\n", d); + + if ((inb(wss_io) & 0x80) != 0) + goto _done; + + if (inb(wss_io + 2) == 0xff) + goto _done; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d); + + if ((inb(wss_io) & 0x80) != 0) + s->type = MEDIA_FX; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { if ((inb(wss_io) & 0x80) == 0) @@ -954,9 +988,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, if (sscape->type == SSCAPE_VIVO) port += 4; - if (dma1 == dma2) - dma2 = -1; - err = snd_wss_create(card, port, -1, irq, dma1, dma2, WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); if (!err) { @@ -1051,21 +1082,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; - - /* - * Check that the user didn't pass us garbage data ... - */ - irq_cfg = get_irq_config(irq[dev]); - if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; - } - - mpu_irq_cfg = get_irq_config(mpu_irq[dev]); - if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; - } + const char *name; /* * Grab IO ports that we will need to probe so that we @@ -1109,8 +1126,41 @@ static int __devinit create_sscape(int dev, struct snd_card *card) goto _release_dma; } - printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n", - sscape->io_base, irq[dev], dma[dev]); + switch (sscape->type) { + case MEDIA_FX: + name = "MediaFX/SoundFX"; + break; + case SSCAPE: + name = "Soundscape"; + break; + case SSCAPE_PNP: + name = "Soundscape PnP"; + break; + case SSCAPE_VIVO: + name = "Soundscape VIVO"; + break; + default: + name = "unknown Soundscape"; + break; + } + + printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", + name, sscape->io_base, irq[dev], dma[dev]); + + /* + * Check that the user didn't pass us garbage data ... + */ + irq_cfg = get_irq_config(sscape->type, irq[dev]); + if (irq_cfg == INVALID_IRQ) { + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + return -ENXIO; + } + + mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); + if (mpu_irq_cfg == INVALID_IRQ) { + printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + return -ENXIO; + } if (sscape->type != SSCAPE_VIVO) { /* @@ -1141,8 +1191,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - activate_ad1845_unsafe(sscape->io_base); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1151,12 +1199,12 @@ static int __devinit create_sscape(int dev, struct snd_card *card) * Enable and configure the DMA channels ... */ sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); - dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40); + dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); - sscape_write_unsafe(sscape->io_base, - GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg); + mpu_irq_cfg |= mpu_irq_cfg << 2; + sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); -- 1.6.0.3 ---------------------------------------------------------------------- Kawalerka za 79 000 zl! Sprawdz >>> http://link.interia.pl/f2348

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[alsa-devel] ASoC update for 2.6.33
by Mark Brown 28 Sep '09

28 Sep '09

The following changes since commit 2c9ee33d37a6f3808c4319f0849671fafa8a6f4c: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Graeme Gregory (1): ASoC: pxa-ssp increase max_channels to 8 sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-)

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[alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 28 Sep '09

28 Sep '09

Tested this on my Zotac IONITX motherboard (N330 model). Dmesg output: [ 16.464533] ALSA hda_codec.c:3081: hda_codec: model '3stack-6ch' is selected for config 174b:437b (Zotac IONITX) Patch: --- alsa-kernel/pci/hda/patch_realtek.c.orig 2009-09-24 14:50:45.000000000 -0700 +++ alsa-kernel/pci/hda/patch_realtek.c 2009-09-24 14:38:07.000000000 -0700 @@ -16724,6 +16724,7 @@ static struct snd_pci_quirk alc662_cfg_t ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x174b, 0x437b, "Zotac IONITX", ALC662_3ST_6ch), {} }; Thanks, Gus ------------------------- Message sent via Atmail Open - http://atmail.org/

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[alsa-devel] No sound after returning from suspend-to-ram
by Ozan Çağlayan 28 Sep '09

28 Sep '09

Hi, A user complains that after returning from suspend-to-ram, he isn't able to get any sound from his laptop until rebooting. I delayed reporting this one because I once thought that it may be related to Pulseaudio or some kind of mixer level problem but now I tried to suspect the kernel and the drivers as well. This is 2.6.30.5 with alsa-* 1.0.21a. I have dmesg and alsa-info outputs before and after s2ram: dmesg before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4454 alsa-info before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4455 dmesg after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4456 alsa-info after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4457 Note that a traceback is dumped in dmesg during resume. Maybe it will give a clue about the problem, dunno. And there's something like hda_codec: invalid CONNECT_LIST verb 12[2]:2100 in regular dmesg output, i wonder what does that mean too. Any ideas? Thanks, Ozan Caglayan

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[alsa-devel] Problem with mplayer and OSS on at91sam9g20(kernel 2.6.30)
by Sedji Gaouaou 28 Sep '09

28 Sep '09

Hi, We've been doing some tests here and we noticed a problem when using mplayer that we didn't have previously(on 2.6.27). While playing audio through OSS emulation on 2.6.30 / at91sam9g20ek there seems to be issue with hardware parameters. The sound does not play and a bunch of error messages are comming on the console. Here is the test case on angstrom: -- mplayer output sample ------------------------- mplayer -quiet myfile.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. [..] AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) [AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-( -- kernel output sample ------------------------- Atmel_ssc_dai: Entering atmel_ssc_hw_params() natmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=4a010404 TFMR=001f018f atmel_ssc_dai,hw_params: SSC initialized asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed [..] asoc: machine hw_params failed MPlayer interrupted by signal 2 in module: ao2_init atmel_ssc_shutdown: transmit disabled SSC_SR=0x000004cc atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback -- 2 error messages to note ------------------------- mplayer: "[AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(" kernel: "asoc: machine hw_params failed" => please note also that it is working correctly using the alsa output: mplayer -ao alsa myfile.mp3 Here is the log while using the alsa output (it is working in this case): root@at91sam:~$ mplayer -quiet -ao alsa /media/sda/audio/01.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. Audio file file format detected. Clip info: Title: Angel Artist: Massive Attack Album: Mezzanine Year: 1998 Comment: Ripped by Beto Track: 1 Genre: Rock ========================================================================== Forced audio codec: mad Opening audio decoder: [libmad] libmad mpeg audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) ========================================================================== atmel_ssc_startup: SSC_SR=0x204 asoc: WM8731 <-> atmel-ssc0 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=19010404 TFMR=001f018f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized transmit enabled SSC_SR=0x000104cf AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... [..] atmel_ssc_shutdown: transmit disabled SSC_SR=0x000104c3 atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown Exiting... (Quit) pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback egards, Sedji

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[alsa-devel] [PATCH 0/1] alsa-lib: pcm_hw: Always use delay ioctl in snd_pcm_delay()
by Kai Vehmanen 28 Sep '09

28 Sep '09

Hello all, this is a updated version of the alsa-lib patch I mentioned in the earlier thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020377.html That was meant to just describe the possible problem, but now I'm more confident that this should be considered for merging. The related alsa-kernel patch is 'ALSA: Add extra delay count in PCM', which added the 'runtime->delay' property. Please review if this patch makes sense. Kai Vehmanen (1): pcm_hw: Always use delay ioctl in snd_pcm_delay() src/pcm/pcm_hw.c | 22 ---------------------- 1 files changed, 0 insertions(+), 22 deletions(-)

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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 28 Sep '09

28 Sep '09

Hi Peter, > Yes it is indicating so. I read through this thread I didn't find any > simple reason why it's not working. Your setup is similar with the > Pandora, i.e. McBSP functional clock is coming the codec and codec is > slave. Yes, I pretty much used the pandora as an example to write mine. > The McBSP is operating if the FSX and CLKX are toggling in this setup. > Is it so that you don't get any DMA interrupts or just few of them? I don't get any DMA int's: ~# cat /proc/interrupts CPU0 11: 0 INTC prcm 12: 0 INTC DMA 25: 0 INTC dispc 37: 5003 INTC gp timer 56: 3425 INTC i2c_omap 62: 0 INTC McBSP 63: 0 INTC McBSP > Your CLKGDV divisor value 256 means too low bit clock and sample rate > for both external 12.288 MHz and internal 96 MHz but still the DMA > should be running (if there is no bug with this divisor value). > Can you try divisor value 8 does it work then? For 48 kHz sample rate > with I2S you need a bit clock of 48 kHz*2*16 = 1.536 MHz and this you > get by dividing the 12.288 MHz with 8. > Yes, I had tried 8 before and just tried it now. Same result. It really seems like there is a DMA issue. Can you think of anything else to try before I dive into the DMA code? Rick

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Re: [alsa-devel] Problems with Intel HDA driver
by Munzir Taha (منذر طه) 27 Sep '09

27 Sep '09

Hi Takashi, Sorry, the correct link to alsa-info report is http://www.alsa-project.org/db/?f=2e50beab11390b9687490feae899217710e81afa On Yaum al-Ithnain 24 Ramadan 1430 4:23:05 pm Takashi Iwai wrote: > At Mon, 14 Sep 2009 15:15:53 +0200, > > I wrote: > > At Mon, 14 Sep 2009 16:15:02 +0300, > > > > Ozan Çağlayan wrote: > > > Takashi Iwai wrote On 14-09-2009 15:36: > > > > Yeah, I think we can add EAPD handling as default for IDT92HD71xx > > > > codecs. I've never seen machines that can't live with GPIO0 bit up. > > > > > > enable_msi doesn't seem to affect the auto-muting problem. I played > > > with every single node in hda-analyzer but can't get auto-mute work. I > > > can manually mute the HP output and the internal speakers by > > > checking/unchecking OUT checkboxes on relevant NIDs. There's a 8xn > > > matrix for GPIO settings, I played with them as well to set unsol but > > > nope. > > > > > > I want to remember that dell-m4-1 makes automute work but breaks the > > > mixer channels: > > > > > > http://www.spinics.net/lists/alsa-devel/msg26714.html > > > > > > Err, I know that I didn't answer the last question in the thread as I'm > > > a little bit lost in the NID pool and didn't have time to investigate > > > further. > > > > Could you check whether this still valid? > > > > Basically the model option doesn't change the auto-mute behavior. > > It just changes which pin to be detected and which pin to mute. > > So, if model=dell-m4-1 makes the auto-muting working, it means > > that the BIOS setup is broken. > > I looked through alsa-info output now, and found the culprit. > NID 0x0f is set as a line-out, and currently the driver doesn't > don't do auto-muting for line-out unless a headphone is present. > Changing it to a headphone should fix. Or you can give hp_detect=1 > hint via sysfs and reconfigure. > > > Takashi -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

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