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[alsa-devel] patch 1/1: ASoC, DMA crash with application- resolved
by naveenkrishna.ch 22 Dec '08

22 Dec '08

Hi ALSA SOC, I am using a OMAP3 + TWL4030 combination with ASoC in the sound/soc/omap/omap-pcm.c DMA channels are being freed after unlinking in the omap_pcm_hw_free function Which is causing a crash during applications So, stopping the DMA channels before unlinking resolves the issue. here is the patch that solves the issue diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index e9084fd..679e384 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -116,6 +116,7 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream) if (prtd->dma_data == NULL) return 0; + omap_stop_dma(prtd->dma_ch); if (!cpu_is_omap1510()) omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); omap_free_dma(prtd->dma_ch); Is this ok or is there any better way out -- Thanks, (: Naveen Krishna Ch :)

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Re: [alsa-devel] [PATCH 0/8]ALSA: ASoc: DaVinci: cleanup
by David Brownell 22 Dec '08

22 Dec '08

On Sunday 21 December 2008, Rajashekhara, Sudhakar wrote: > the existing meaning is correct w.r.t davinci McBSP. Not entirely. DaVinci doesn't have McBSP; it has ASP. Having all this code refer to a McBSP -- as found for example on OMAP chips -- is confusing and incorrect. - Dave

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[alsa-devel] Sigmatel codec on Dell Vostro 1400
by Paulo Cavalcanti 21 Dec '08

21 Dec '08

Hi, I have a Dell Vostro 1400 laptop, which uses the sigmatel codec STAC9228. This laptop has a digital built in mic and a connector for an external analog mic. It also has two headphones connectors. Using yesterday's alsa tarball model=ref or ref-no-jd - sound in headphones only, external analog mic works. No digital mic. model=3stack,5stack - no sound at all. model=dell-3stack - sound works for headphones and speakers (jack detection mutes speakers when headphones are inserted), no analog external mic, and digital mic works, but sound is really too low. Therefore, I would be very pleased if I could either use the analog mic with model=dell-3stack, or if the built in digital mic had a louder sound. Using hda-verb-0.3, the pins are: 0xa - headphones (first connector - the one which works) 0xf - headphones (second connector - mute) 0xe - analog mic The information from alsa-info is here: http://www.alsa-project.org/db/?f=6a64faf3cd93c8c805e868c33eff88aec29508c8 Thanks. -- Paulo Roma Cavalcanti LCG - UFRJ

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[alsa-devel] Additing a new Hardware Specific feature in the ALSA driver
by SUBHRANIL CHOUDHURY 21 Dec '08

21 Dec '08

Dear all, I am new to ALSA programming. Suppose if i have to add some hardware specific feature in the ALSA sound driver and expose to the application via the ALSA library, how can do that? Please reply if you have some inputs. Anything is welcome. Thanks in advance. Regards, Subhranil

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[alsa-devel] [PATCH] hda: fix incorrect mixer index values for 92hd83xx
by Matthew Ranostay 21 Dec '08

21 Dec '08

Fixed incorrect mixer index values for 92hd83xx codec's audio input mixer. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 70181d5..efa1fb0 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -1101,21 +1101,21 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT), - HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x2, HDA_INPUT), /* - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x1, HDA_INPUT), */ { } /* end */ };

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[alsa-devel] [PATCH] hda: dinput_mux check
by Matthew Ranostay 21 Dec '08

21 Dec '08

Add check to determine if dinput_mux is set by any of patch_stac*() functions, otherwise a invalid pointer my be referenced causing gibberish to mixer values. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 70181d5..fa87893 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -3601,7 +3601,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; + if (!spec->dinput_mux) + spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; spec->amp_mux = &spec->private_amp_mux;

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[alsa-devel] somebody ever see ALSA applied in robotics ?
by Ronald 20 Dec '08

20 Dec '08

Hi, all, I'm trying to develop a system which can determine the direction(s) sound comes from. In the end, it will be put in an R/c submarine, on a gumstix linux board (see http://www.subcommittee.com for my hobby) But for now, I develop in air, with ubuntu. all I want is to record using both the mic channel and the line in (where another 2 mics are attached with a small preamp), and I want to record while emitting a chirp, analyzing echo's. the concept has been shown to work. see for instance http://eddiem.com/projects/chirp/chirp.htm I am assuming that since all this goes via the same sound card, the same crystal is used, and samples from mic and line in are essentially synchronized (I only need it to be sync-ish for < 0,1 sec..) Now I have read a lot of ALSA tutorials and such like, but I'm lost. a lot is about humans consuming nice harmonics, sharing and mixing, which is not applicable to my case. I have the nagging feeling OSS is a better fit for my case, but OSS is apparently on a 60 day licence now. And, I am a newbie coder. no surprise there ;-) So, these are my questions : - Do I need to combine mic and line-in into one device using the config file ? do I get a 3 channel device then ? does a plug-in add latency ? - can I sync the transmission of a short chirp with the start of the recording on these 3 channels ? is starting them close after each other in the program OK, or is there possibly large latency caused *after* I call the API ? the "real" sync'ing will be the (first) recording of the chirp once it leaves the speaker, so it isn't critical, but I would like to know because it might save a few bytes useless samples. - Does anybody out there have any links, code snippets or general published info on how to apply ALSA to the robotics field ? Ps I have read the el-cheapo hack with multiple soundcards, and while a very good idea, I cannot use it in the boat later on.. thanks for any input you might have. Regards, Ronald van Aalst

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[alsa-devel] Unknown symbols in snd-fm801.ko
by Ozan Çağlayan 20 Dec '08

20 Dec '08

Hi, Yesterday I was trying to compile the latest snapshot of alsa-driver on top of kernel 2.6.28_rc8. When looking through the console output of make, I saw two warnings given by LD reporting two unknown symbols in snd-fm801.ko(driver for a ForteMedia sound card): Building modules, stage 2. MODPOST 170 modules WARNING: "snd_tea575x_init" [/var/pisi/alsa-driver-1.0.18a_20081214-39/work/alsa-driver/pci/snd-fm801.ko] undefined! WARNING: "snd_tea575x_exit" [/var/pisi/alsa-driver-1.0.18a_20081214-39/work/alsa-driver/pci/snd-fm801.ko] undefined! laptop alsa-driver # grep -rHIn "snd_tea575x_init" * alsa-kernel/i2c/other/tea575x-tuner.c:195:void snd_tea575x_init(struct snd_tea575x *tea) alsa-kernel/i2c/other/tea575x-tuner.c:251:EXPORT_SYMBOL(snd_tea575x_init); alsa-kernel/include/tea575x-tuner.h:51:void snd_tea575x_init(struct snd_tea575x *tea); alsa-kernel/pci/fm801.c:1447: snd_tea575x_init(&chip->tea); i2c/other/tea575x-tuner.c:236:void snd_tea575x_init(struct snd_tea575x *tea) i2c/other/tea575x-tuner.c:309:EXPORT_SYMBOL(snd_tea575x_init); include/sound/tea575x-tuner.h:51:void snd_tea575x_init(struct snd_tea575x *tea); pci/fm801.c:1448: snd_tea575x_init(&chip->tea); pci/fm801.c.orig:1447: snd_tea575x_init(&chip->tea); sound/i2c/other/tea575x-tuner.c:195:void snd_tea575x_init(struct snd_tea575x *tea) sound/i2c/other/tea575x-tuner.c:251:EXPORT_SYMBOL(snd_tea575x_init); sound/include/tea575x-tuner.h:51:void snd_tea575x_init(struct snd_tea575x *tea); sound/pci/fm801.c:1447: snd_tea575x_init(&chip->tea); So this symbols are exported by tea575x-tuner.c. There's a card option parameter in configuration of alsa-driver called 'fm801-tea575x-bool'.But eventhough, I'm configuring the package with, "--with-card-options=all", I'm getting this warnings. As a consequence it's impossible to load the module snd-fm801. I think that it's a problem with Kconfig stuff but I really don't get how alsa-driver package manages the whole thing with alsa-kernel directory, patchsets etc.(BTW, is there any documentation about the mechanism in alsa-driver, and the functions of separate git repos alsa-driver, alsa-kernel, stub, etc.?) I'm having the issue with 2.6.25.x && (1.0.17 || 1.0.18), 2.6.28_rc8 && (1.0.18) so the problem may not be caught until this time because there exists a few distributions which ships alsa-driver(and other kernel modules) separately. Regards, -- Ozan Çağlayan <ozan_at_pardus.org.tr>

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[alsa-devel] ice1712 error
by Udo van den Heuvel 20 Dec '08

20 Dec '08

Hello, I found out the ICE1712 card is sometimes detected OK and sometimes isn't detected correctly. Cold boot/warm boot doesn't really matter. I tried kernel 2.6.27.9 and older. Card is is fine when detected. Behaviour doesn't change when I try a different PCI slot. When the card fails: # dmesg|grep 01:09 PCI: 0000:01:09.0 reg 10 io port: [a000, a01f] PCI: 0000:01:09.0 reg 14 io port: [a400, a40f] PCI: 0000:01:09.0 reg 18 io port: [a800, a80f] PCI: 0000:01:09.0 reg 1c io port: [ac00, ac3f] pci 0000:01:09.0: supports D2 ICE1712: probe of 0000:01:09.0 failed with error -12 When the card is OK: PCI: 0000:01:09.0 reg 10 io port: [a000, a01f] PCI: 0000:01:09.0 reg 14 io port: [a400, a40f] PCI: 0000:01:09.0 reg 18 io port: [a800, a80f] PCI: 0000:01:09.0 reg 1c io port: [ac00, ac3f] pci 0000:01:09.0: supports D2 ICE1712 0000:01:09.0: PCI INT A -> Link[APC4] -> GSI 19 (level, low) -> IRQ 19 I booted into single user mode each time and checked /proc/asound/cards after each boot and then looked at dmesg. Does this mean the card is toast? What does the -12 mean? What should I try next? Kind regards, Udo

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[alsa-devel] [PATCHv2 1/1] Fix a Oops bug in omap soc driver.
by Stanley.Miao 20 Dec '08

20 Dec '08

There will be a Oops or frequent underrun messages when playing music with omap soc driver, this is because a data region is incorretly sized, other data region will be overwriten when writing to this data region. Signed-off-by: Stanley.Miao <stanley.miao(a)windriver.com> --- sound/soc/omap/omap-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index ce580a9..2adfbce 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -233,7 +233,7 @@ static int omap_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) goto out; - prtd = kzalloc(sizeof(prtd), GFP_KERNEL); + prtd = kzalloc(sizeof(struct omap_runtime_data), GFP_KERNEL); if (prtd == NULL) { ret = -ENOMEM; goto out; -- 1.5.6.3

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