December 2008 - Alsa-devel - mailman.alsa-project.org

[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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[alsa-devel] ALsa Querries about buffer size configuration
by SUBHRANIL CHOUDHURY 20 Dec '08

20 Dec '08

I have some doubts on the ALSA programming, 1)My ALSA driver supports minimum Period Bytes size as 32 bytes, but when i set the period size as 512 bytes it is always setting to 1024 bytes.I am using the function "snd_pcm_hw_params_set_period_size_near()" . What might be wrong here?Is the function what i am using correct? 2)How the Library functions and the Driver functions interact, what is the interface between the library functions and driver functions?Is there any document which explains this. I got a document which explains how to write ALSA driver.But it does not mention the interface with the library. 3)How Can we achive MIC and LINEIN switching using the mixer?What library function do i need to use to achive this? 4)If i want to get the Timestamp for each buffer captured, what library function should be used? 5)I am trying to capture and play the same audio (ALSA duplex). When i capture and store it in a file and play from the file i am able to hear the sound properly. If i capture in a buffer and play the same buffer directly it says Broken PIPE ERROR in the playout. I tried the capture and playout in two different threads, i am able to hear the sound and i am not getting any Broken PIPE error. But along with the audio i am hearing some noise. Do we need to use some kind of synchronisation between the threads for capture and playout? Any help on these things is appreciated. Thank you in advance. Regards, Subhranil

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[alsa-devel] Additing a new Hardware Specific feature in the ALSA driver
by SUBHRANIL CHOUDHURY 20 Dec '08

20 Dec '08

On Sat, Dec 20, 2008 at 7:41 PM, SUBHRANIL CHOUDHURY <subhra85(a)gmail.com>wrote: > > > 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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Re: [alsa-devel] [PATCH 01/01] alsa: sound/soc/pxa/pxa2xx_pcm.c, check and set invalid dma_ch to be <= to 0 in pxa2xx_pcm_hw_free()
by Mark Brown 20 Dec '08

20 Dec '08

On Fri, Dec 19, 2008 at 09:52:22PM -0800, sware wrote: > From: Stephen Ware <stephen.ware(a)eqware.net>, Dennis O'Brien <dennis.obrien(a)eqware.net> ... > Signed-off-by: Dennis O'Brien <dennis.obrien(a)eqware.net> The patch itself is good but could you please clarify what's going on with the author and signoff lines (apart from anything else the multiple authors appears to be confusing git, it's ignoring Dennis). The combination of git format-patch and send-email is handy for generating something that applies easily.

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[alsa-devel] [PATCH 1/4] ALSA: ASoC: tlv320aic3x add dsp_a
by Troy Kisky 20 Dec '08

20 Dec '08

Add SND_SOC_DAIFMT_DSP_A mode option. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b76bcc3..b9b67ed 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -847,6 +847,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; u8 iface_areg, iface_breg; + int delay = 0; iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; @@ -872,6 +873,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, SND_SOC_DAIFMT_INV_MASK)) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): + delay = 1; case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; @@ -888,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set iface */ aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00a195a..40a6c2a 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -35,6 +35,8 @@ #define AIC3X_ASD_INTF_CTRLA 8 /* Audio serial data interface control register B */ #define AIC3X_ASD_INTF_CTRLB 9 +/* Audio serial data interface control register C */ +#define AIC3X_ASD_INTF_CTRLC 10 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 /* Audio codec digital filter control register */ -- 1.5.4.3

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[alsa-devel] "No codecs found" on Intel HDA + AD1988A
by Alexander Chemeris 19 Dec '08

19 Dec '08

Hi, I have Asus P5B-V motherboard with integrated Intel HDA audio and AD1988A codec (I see the chip on the board). About half a year ago or so it used to work, but now when snd-hda-intel module is loaded it claims it can't find a codec: $ dmesg | grep -i hda [ 13.648205] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 13.648236] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 13.680537] hda-intel: no codecs found! [ 13.680589] HDA Intel 0000:00:1b.0: PCI INT A disabled I tried different recent versions of ALSA to no avail. So far I've tried: * 1.0.17 with Ubuntu patches * 1.0.18 with Ubuntu patches * 1.0.18a downloaded half an hour ago from here: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… I also tried to compile 1.0.16, but seems it is not compatible with the kernel: $ uname -a Linux nidus 2.6.27-10-generic #1 SMP Fri Nov 21 19:19:18 UTC 2008 x86_64 GNU/Linux When I tried 1.0.18a, I also manually loaded snd-hda-codec_analog before loading snd-hda-intel without any success. And yeah, when I just run `modprobe snd-hda-intel` only following modules are loaded: $ lsmod | grep snd snd_hda_intel 37864 0 snd_hda_codec 86784 1 snd_hda_intel snd_hwdep 17160 1 snd_hda_codec snd_pcm_oss 52256 0 snd_mixer_oss 25088 1 snd_pcm_oss snd_pcm 98824 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_oss 43776 0 snd_seq_midi_event 16768 1 snd_seq_oss snd_seq 67744 4 snd_seq_oss,snd_seq_midi_event snd_timer 33424 2 snd_pcm,snd_seq snd_seq_device 16788 2 snd_seq_oss,snd_seq snd 83144 10 snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device soundcore 16800 1 snd snd_page_alloc 18576 2 snd_hda_intel,snd_pcm What can I do to help get my audio working again? I don't want to install different kernel or restart computer (well, as a last resort, may be...), but I easily can test different any patches, etc. PS I don't think it matters, but I use Ubuntu 8.10 x64. -- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000

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Re: [alsa-devel] No HDMI audio on Asus P5Q-EM
by Chris Pimlott 19 Dec '08

19 Dec '08

Forgot to join the list first, also here is the bugtrack issue: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4281 On Thu, Dec 11, 2008 at 3:41 PM, Chris Pimlott <pimlottc(a)gmail.com> wrote: > I have an Asus P5Q-EM motherboard, with Intel G45 chipset and Realtek ALC > 1200 integrated audio, with HDMI output. (Specifications: > http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…)<http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…>[ > ^<http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…> > ] > > The analog output has always worked fine (from the default Mythbuntu > install), but I do not get any output when trying to use the HDMI. I am > testing with aplay: "aplay -D plughw:0,3 > /usr/share/sounds/alsa/Front_Center.wav". plughw:0,1 with the same command > works (on analog out). > > I see three devices, "IEC958", "IEC958 Default PCM" and "IEC958 1", which > are all unmuted. I have tried flipped the copyright setting with "iecset -n > 1 copy on" but that has made no difference. > > I have tried a number of versions, including the stock 2.6.27 kernel > package shipped in Mythbuntu, alsa-1.0.18 and alsa-1.0.18a, as well as > version of the sound-2.6 from 2008-11-23 and today. The most recent I'm > trying is sound-2.6 version is b13a9d80db0f1f4ace2863e8824a3d3299730d0a. > > I've notice there's been a fair amount of updates to this area of the > snd_hda_intel driver, and some changes to how the device has shown up in > recent releases. Should I even expect this to work, or is it still > "in-progress"? Thanks. > > Chris Pimlott >

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[alsa-devel] arecord ..xrun!!!!
by vasudha rao 19 Dec '08

19 Dec '08

Hi .. I am trying to record .wav files.. I did like, $arecord -D copy test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono xrun!!! (at least 0.251 ms long) xrun!!! (at least 0.071 ms long) xrun!!! (at least 0.072 ms long) . . . When I run this test.wav, no sound!!!! Am I doing wrong anywhere?? Thanks:-)

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[alsa-devel] arecord working now..
by vasudha rao 19 Dec '08

19 Dec '08

Hi.. I am able to record now.. I used arecord like, $arecord -d 10 -f cd -t wav -D copy foobar.wav and I play like, $aplay -D hw:0,0 foobar.wav All works fine..:-) But the thing is I dont know why it was not working and why it is working now..?? Please tel me if u found it why??and if time permits for u.. Thank you ..

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Re: [alsa-devel] Patch to /sound/soc/pxa/pxa2xx_pcm.c, WHO IS INTERESTED?
by Mark Brown 19 Dec '08

19 Dec '08

On Thu, Dec 18, 2008 at 10:34:53AM -0800, sware wrote: > Patch is to 2.6.27.9 (yesterday), notice .10 is up today, > patch should still apply. Thanks for your patch. Unfortunately all this code has been moved about a lot in the latest code in order to share the code with the non-ASoC AC97 driver so your patch won't apply - see the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for the latest code. Could you rebase your work against that, please? Also, please include a Signed-off-by line for patch submissions - for details see Documentation/SubmittingPatches > diff -uprN -X linux-2.6.27.9.orig/Documentation/dontdiff linux-2.6.27.9.orig/sound/soc/pxa/pxa2xx-pcm.c linux-2.6.27.9.snd_modified/sound/soc/pxa/pxa2xx-pcm.c > --- linux-2.6.27.9.orig/sound/soc/pxa/pxa2xx-pcm.c 2008-12-13 15:56:16.000000000 -0800 > +++ linux-2.6.27.9.snd_modified/sound/soc/pxa/pxa2xx-pcm.c 2008-12-17 14:09:02.000000000 -0800 > @@ -137,13 +137,18 @@ static int pxa2xx_pcm_hw_free(struct snd > { > struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; > > - if (prtd && prtd->params) > + if (!prtd) > + return 0; > + if (!prtd->params) > + return 0; > + prtd should always be there - the code should at least complain loudly if it's NULL. params might potentially be hit, though. > *prtd->params->drcmr = 0; > > - if (prtd->dma_ch) { > + if (prtd->dma_ch >= 0) { > snd_pcm_set_runtime_buffer(substream, NULL); > pxa_free_dma(prtd->dma_ch); > - prtd->dma_ch = 0; > + prtd->dma_ch = -1; /* invalid channel */ > + prtd->params = NULL; > } This bit is good. > @@ -266,6 +271,8 @@ static int pxa2xx_pcm_close(struct snd_p > dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, > prtd->dma_desc_array, prtd->dma_desc_array_phys); > kfree(prtd); > + runtime->private_data = NULL; > + > return 0; > } As is this.

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[alsa-devel] Streams
by James Courtier-Dutton 19 Dec '08

19 Dec '08

Hi, There was some talk some time ago of maybe moving alsa to a streams approach instead of ring buffer and periods approach. Has anything become of this? Is there a plan to improve the abstraction between hardware and user space regarding constraints on period sizes when doing sample rate conversion? Kind Regards James

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[alsa-devel] 44.1khz capture on CA0106?
by Takashi Iwai 19 Dec '08

19 Dec '08

Hi, can anyone confirm that 44.1kHz capture on CA0106 causes the problem with 48kHz playback? Typically, recording via 44.kHz using hw device, then run 48kHz playback. % arecord -fcd -Dhw:0 foo.wav % aplay -Dhw:0 48khz-samples.wav I got a bug report about this, and disabling 44.1kHz capture fixes the problem. His case is with MSI K8N mobo, and I'd like to know whether this happens in general on CA0106. The bug report is found in https://bugzilla.novell.com/show_bug.cgi?id=447624 thanks, Takashi

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[alsa-devel] ca0106 SPDIF defaults
by Ben Stanley 19 Dec '08

19 Dec '08

Hi, I previously reported problems that are still plaguing me [1] to do with SPDIF settings that don't reset properly. I am now looking at doing a patch to the ca0106 driver to fix this problem. It appears I have to add a new control to be able to set the default settings. I presume this introduces complications with existing settings in asound.state? Is there some collected wisdom on how to add new controls while providing a migration/upgrade/interoperability path? Ben Stanley. [1] http://article.gmane.org/gmane.linux.alsa.devel/55406/match=independent+pcm…

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[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 19 Dec '08

19 Dec '08

The following changes since commit 49d92c7d5bbd158734bc34ed578a68b214a48583: Stanley.Miao (1): ASoC: TWL4030: hands-free start-up sequence. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Alexander Beregalov (1): ASoC: switch davinci DPRINTK to pr_debug() Mark Brown (3): ASoC: Ease merge difficulties from new architectures ASoC: Complain if we fail to create DAPM controls ASoC: Add WM8350 AudioPlus codec driver include/linux/mfd/wm8350/audio.h | 38 +- sound/soc/Kconfig | 10 +- sound/soc/Makefile | 12 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8350.c | 1583 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 20 + sound/soc/davinci/davinci-pcm.c | 18 +- sound/soc/soc-dapm.c | 6 +- 9 files changed, 1669 insertions(+), 24 deletions(-) create mode 100644 sound/soc/codecs/wm8350.c create mode 100644 sound/soc/codecs/wm8350.h

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[alsa-devel] [ASoC] TWL4030 underrun error
by naveenkrishna.ch 18 Dec '08

18 Dec '08

Hello, I am working on OMAP3 + TWL4030 combination for ASoC codec The omap generic omap-pcm.c file has static const struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_96000, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, .periods_max = 255, .buffer_bytes_max = 128 * 1024, }; following structure, When i play with aplay XXX.wav Rare but i used to get some underrun messages i found thses message are coming from aplay. underrun!!! (at least 10271.545 ms long) underrun!!! (at least 904.663 ms long) underrun!!! (at least 771.790 ms long) underrun!!! (at least 3393.433 ms long) underrun!!! (at least 777.314 ms long) I dint found much info about this in net too. How can i solve this, can any one help me out -- Thanks, (: Naveen Krishna Ch :)

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[alsa-devel] [PATCH] ALSA: Add support for TV-out to the jack reporting API
by Jani Nikula 18 Dec '08

18 Dec '08

Add support for reporting new jack types SND_JACK_TVOUT and SND_JACK_AVOUT (a combination of LINEOUT and TVOUT) to the jack reporting API. Also add the corresponding SW_TVOUT_INSERT switch to the input system header. Signed-off-by: Jani Nikula <ext-jani.1.nikula(a)nokia.com> --- include/linux/input.h | 1 + include/sound/jack.h | 2 ++ sound/core/jack.c | 6 ++++++ 3 files changed, 9 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index abd223b..ea5b224 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -646,6 +646,7 @@ struct input_absinfo { #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ +#define SW_TVOUT_INSERT 0x08 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 2e0315c..69fc1f0 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -37,6 +37,8 @@ enum snd_jack_types { SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ + SND_JACK_TVOUT = 0x0010, + SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_TVOUT, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index 7ca905c..23c3f3e 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -105,6 +105,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_LINEOUT) input_set_capability(jack->input_dev, EV_SW, SW_LINEOUT_INSERT); + if (type & SND_JACK_TVOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_TVOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -162,6 +165,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_LINEOUT) input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, status & SND_JACK_LINEOUT); + if (jack->type & SND_JACK_TVOUT) + input_report_switch(jack->input_dev, SW_TVOUT_INSERT, + status & SND_JACK_TVOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); -- 1.6.0.4

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[alsa-devel] aplay..no sound..no error..
by vasudha rao 17 Dec '08

17 Dec '08

Hi all.. I have installed alsa for my embedded board..(at91sam9263) when I use aplay utility I got, $ aplay a.wav Playing WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono I am not getting any error message.. But no sound from speaker also.. What else I can try to listen sound from speaker..?? Thanks,

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[alsa-devel] SND_SOC_DAIFMT_NB_NF meaning
by Troy Kisky 17 Dec '08

17 Dec '08

A quick question here. Does SND_SOC_DAIFMT_NB_NF have a standard meaning or does it depend on what mode your in (I2S, DSP_A, right just). From looking at the sources, SND_SOC_DAIFMT_NB_NF usually means that data is valid on the rising BCLK and start of frame is rising edge if DSP mode. The reason I ask, is that I'd like to pass the same flags to the codec and the cpu_dai and have them agree with the interpretation. Thanks Troy

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