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Re: [alsa-devel] [PATCH] ASoC: typo
by Liam Girdwood 22 Dec '08

22 Dec '08

On Mon, 2008-12-22 at 17:40 +0100, Roel Kluin wrote: > Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> > --- > diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c > index acd68ef..227c865 100644 > --- a/sound/soc/omap/omap-pcm.c > +++ b/sound/soc/omap/omap-pcm.c > @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, > prtd->dma_data = dma_data; > err = omap_request_dma(dma_data->dma_req, dma_data->name, > omap_pcm_dma_irq, substream, &prtd->dma_ch); > - if (!err & !cpu_is_omap1510()) { > + if (!err && !cpu_is_omap1510()) { > /* > * Link channel with itself so DMA doesn't need any > * reprogramming while looping the buffer Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Liam

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[alsa-devel] ASoC updaets
by Mark Brown 22 Dec '08

22 Dec '08

The following changes since commit a31501d1041c9d0a6c3f520736ae2b2fa081493a: Takashi Iwai (1): ALSA: ASoC - Add missing __devexit annotation to wm8350.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Jarkko Nikula (2): ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers ASoC: Fix DSP formats in SSM2602 audio codec sound/soc/codecs/ssm2602.c | 4 ++-- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/omap/omap-mcbsp.c | 4 ++-- sound/soc/omap/osk5912.c | 4 ++-- 4 files changed, 7 insertions(+), 7 deletions(-)

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[alsa-devel] [topic/asoc][PATCH 0/1] ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers
by Jarkko Nikula 22 Dec '08

22 Dec '08

Hi This fix is generated against topic/asoc and don't apply against mainline. IMO this can go to next merge window with preceding patches since OSK5912 in mainline should work, just format is defined wrong and all new drivers where this affects should anyway go to the top of topic/asoc. -- Jarkko

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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 20 Dec '08

20 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 20 Dec '08

20 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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