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

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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[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

2 1

[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

1 0

[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…

2 1

[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

3 6

[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 :)

3 2

[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

3 8

[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,

3 2

[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

2 1

Re: [alsa-devel] 1.0.18: /proc/asound/card0/stream0 no longer existing?
by Takashi Iwai 17 Dec '08

17 Dec '08

At Wed, 17 Dec 2008 10:49:08 +0100, Klaus Schulz wrote: > > Takashi. > > As you can see below SND_VERBOSE_PROCFS is y. But DEBUG and VERBOSE_PRINTK is > NO! These are irrelevant. > BTW: I am running a 2.6.28-rc6-zen1 kernel. I don't know what it means. But the stream* proc file should be created for snd-usb-audio when CONFIG_PROC_FS and CONFIG_SND_VERBOSE_PROCFS are set. At least, there is no change in the ALSA upstream side. Takashi > > Cheers > Klaus > > CONFIG_SOUND=m > CONFIG_SOUND_OSS_CORE=y > CONFIG_SND=m > CONFIG_SND_TIMER=m > CONFIG_SND_PCM=m > CONFIG_SND_HWDEP=m > CONFIG_SND_RAWMIDI=m > CONFIG_SND_SEQUENCER=m > CONFIG_SND_SEQ_DUMMY=m > CONFIG_SND_OSSEMUL=y > CONFIG_SND_MIXER_OSS=m > CONFIG_SND_PCM_OSS=m > CONFIG_SND_PCM_OSS_PLUGINS=y > CONFIG_SND_SEQUENCER_OSS=y > CONFIG_SND_DYNAMIC_MINORS=y > CONFIG_SND_SUPPORT_OLD_API=y > CONFIG_SND_VERBOSE_PROCFS=y > # CONFIG_SND_VERBOSE_PRINTK is not set > # CONFIG_SND_DEBUG is not set > CONFIG_SND_VMASTER=y > CONFIG_SND_MPU401_UART=m > CONFIG_SND_OPL3_LIB=m > CONFIG_SND_OPL4_LIB=m > CONFIG_SND_VX_LIB=m > CONFIG_SND_AC97_CODEC=m > CONFIG_SND_DRIVERS=y > CONFIG_SND_PCSP=m > CONFIG_SND_DUMMY=m > CONFIG_SND_VIRMIDI=m > CONFIG_SND_MTPAV=m > CONFIG_SND_MTS64=m > CONFIG_SND_SERIAL_U16550=m > CONFIG_SND_MPU401=m > CONFIG_SND_PORTMAN2X4=m > CONFIG_SND_AC97_POWER_SAVE=y > CONFIG_SND_AC97_POWER_SAVE_DEFAULT=0 > CONFIG_SND_WSS_LIB=m > CONFIG_SND_SB_COMMON=m > CONFIG_SND_SB8_DSP=m > CONFIG_SND_SB16_DSP=m > CONFIG_SND_ISA=y > CONFIG_SND_ADLIB=m > CONFIG_SND_AD1816A=m > CONFIG_SND_AD1848=m > CONFIG_SND_ALS100=m > CONFIG_SND_AZT2320=m > CONFIG_SND_CMI8330=m > CONFIG_SND_CS4231=m > CONFIG_SND_CS4232=m > CONFIG_SND_CS4236=m > CONFIG_SND_DT019X=m > CONFIG_SND_ES968=m > CONFIG_SND_ES1688=m > CONFIG_SND_ES18XX=m > CONFIG_SND_SC6000=m > CONFIG_SND_GUSCLASSIC=m > CONFIG_SND_GUSEXTREME=m > CONFIG_SND_GUSMAX=m > CONFIG_SND_INTERWAVE=m > CONFIG_SND_INTERWAVE_STB=m > CONFIG_SND_OPL3SA2=m > CONFIG_SND_OPTI92X_AD1848=m > CONFIG_SND_OPTI92X_CS4231=m > CONFIG_SND_OPTI93X=m > CONFIG_SND_MIRO=m > CONFIG_SND_SB8=m > CONFIG_SND_SB16=m > CONFIG_SND_SBAWE=m > CONFIG_SND_SB16_CSP=y > CONFIG_SND_SGALAXY=m > CONFIG_SND_SSCAPE=m > CONFIG_SND_WAVEFRONT=m > CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL=y > CONFIG_SND_PCI=y > CONFIG_SND_AD1889=m > CONFIG_SND_ALS300=m > CONFIG_SND_ALS4000=m > CONFIG_SND_ALI5451=m > CONFIG_SND_ATIIXP=m > CONFIG_SND_ATIIXP_MODEM=m > CONFIG_SND_AU8810=m > CONFIG_SND_AU8820=m > CONFIG_SND_AU8830=m > CONFIG_SND_AW2=m > CONFIG_SND_AZT3328=m > CONFIG_SND_BT87X=m > # CONFIG_SND_BT87X_OVERCLOCK is not set > CONFIG_SND_CA0106=m > CONFIG_SND_CMIPCI=m > CONFIG_SND_OXYGEN_LIB=m > CONFIG_SND_OXYGEN=m > CONFIG_SND_CS4281=m > CONFIG_SND_CS46XX=m > CONFIG_SND_CS46XX_NEW_DSP=y > CONFIG_SND_CS5530=m > CONFIG_SND_CS5535AUDIO=m > CONFIG_SND_DARLA20=m > CONFIG_SND_GINA20=m > CONFIG_SND_LAYLA20=m > CONFIG_SND_DARLA24=m > CONFIG_SND_GINA24=m > CONFIG_SND_LAYLA24=m > CONFIG_SND_MONA=m > CONFIG_SND_MIA=m > CONFIG_SND_ECHO3G=m > CONFIG_SND_INDIGO=m > CONFIG_SND_INDIGOIO=m > CONFIG_SND_INDIGODJ=m > CONFIG_SND_EMU10K1=m > CONFIG_SND_EMU10K1X=m > CONFIG_SND_ENS1370=m > CONFIG_SND_ENS1371=m > CONFIG_SND_ES1938=m > CONFIG_SND_ES1968=m > CONFIG_SND_FM801=m > CONFIG_SND_FM801_TEA575X_BOOL=y > CONFIG_SND_FM801_TEA575X=m > CONFIG_SND_HDA_INTEL=m > # CONFIG_SND_HDA_HWDEP is not set > # CONFIG_SND_HDA_INPUT_BEEP is not set > CONFIG_SND_HDA_CODEC_REALTEK=y > CONFIG_SND_HDA_CODEC_ANALOG=y > CONFIG_SND_HDA_CODEC_SIGMATEL=y > CONFIG_SND_HDA_CODEC_VIA=y > CONFIG_SND_HDA_CODEC_ATIHDMI=y > CONFIG_SND_HDA_CODEC_NVHDMI=y > CONFIG_SND_HDA_CODEC_CONEXANT=y > CONFIG_SND_HDA_CODEC_CMEDIA=y > CONFIG_SND_HDA_CODEC_SI3054=y > CONFIG_SND_HDA_GENERIC=y > CONFIG_SND_HDA_POWER_SAVE=y > CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0 > CONFIG_SND_HDSP=m > CONFIG_SND_HDSPM=m > CONFIG_SND_HIFIER=m > CONFIG_SND_ICE1712=m > CONFIG_SND_ICE1724=m > CONFIG_SND_INTEL8X0=m > CONFIG_SND_INTEL8X0M=m > CONFIG_SND_KORG1212=m > CONFIG_SND_MAESTRO3=m > CONFIG_SND_MIXART=m > CONFIG_SND_NM256=m > CONFIG_SND_PCXHR=m > CONFIG_SND_RIPTIDE=m > CONFIG_SND_RME32=m > CONFIG_SND_RME96=m > CONFIG_SND_RME9652=m > CONFIG_SND_SIS7019=m > CONFIG_SND_SONICVIBES=m > CONFIG_SND_TRIDENT=m > CONFIG_SND_VIA82XX=m > CONFIG_SND_VIA82XX_MODEM=m > CONFIG_SND_VIRTUOSO=m > CONFIG_SND_VX222=m > CONFIG_SND_YMFPCI=m > CONFIG_SND_SPI=y > CONFIG_SND_USB=y > CONFIG_SND_USB_AUDIO=m > CONFIG_SND_USB_USX2Y=m > CONFIG_SND_USB_CAIAQ=m > CONFIG_SND_USB_CAIAQ_INPUT=y > # CONFIG_SND_USB_US122L is not set > CONFIG_SND_PCMCIA=y > CONFIG_SND_VXPOCKET=m > CONFIG_SND_PDAUDIOCF=m > CONFIG_SND_SOC=m > # CONFIG_SND_SOC_ALL_CODECS is not set > CONFIG_SOUND_PRIME=m > CONFIG_SOUND_MSNDCLAS=m > CONFIG_MSNDCLAS_INIT_FILE="/etc/sound/msndinit.bin" > CONFIG_MSNDCLAS_PERM_FILE="/etc/sound/msndperm.bin" > CONFIG_SOUND_MSNDPIN=m > CONFIG_MSNDPIN_INIT_FILE="/etc/sound/pndspini.bin" > CONFIG_MSNDPIN_PERM_FILE="/etc/sound/pndsperm.bin" > CONFIG_SOUND_OSS=m > # CONFIG_SOUND_TRACEINIT is not set > CONFIG_SOUND_DMAP=y > CONFIG_SOUND_SSCAPE=m > CONFIG_SOUND_VMIDI=m > CONFIG_SOUND_TRIX=m > CONFIG_SOUND_MSS=m > CONFIG_SOUND_MPU401=m > CONFIG_SOUND_PAS=m > CONFIG_SOUND_PSS=m > CONFIG_PSS_MIXER=y > # CONFIG_PSS_HAVE_BOOT is not set > CONFIG_SOUND_SB=m > CONFIG_SOUND_YM3812=m > CONFIG_SOUND_UART6850=m > CONFIG_SOUND_AEDSP16=m > CONFIG_SC6600=y > CONFIG_SC6600_JOY=y > CONFIG_SC6600_CDROM=4 > CONFIG_SC6600_CDROMBASE=0 > CONFIG_SOUND_KAHLUA=m > CONFIG_AC97_BUS=m > CONFIG_HID_SUPPORT=y > CONFIG_HID=m > # CONFIG_HID_DEBUG is not set > CONFIG_HIDRAW=y > > On Wed, Dec 17, 2008 at 9:40 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 17 Dec 2008 08:37:58 +0100, > Klaus Schulz wrote: > > > > Hi there. > > > > Can anybody tell me if /proc/asound/card0/stream0 is not existing any > longer > > and what has replaced it? > > It's built only when kernel config CONFIG_SND_VERBOSE_PROCFS=y. > In the case of alsa-driver tarball, this is enabled unless you add > --disable-verbose-procfs configure option. > > Takashi > >

1 0

[alsa-devel] Sigmatel patch for missing
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi folks. Below patch seems to be missing in 1.0.18a! Note that also MCP51 seemed to work afterwards. It needs to be ported to 1.0.18.a though. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3036 Can anybody let me know how the patch would look like in latest driver rev. THX Klaus --- patch_sigmatel.c.orig 2007-09-15 17:57:07.000000000 -0700 +++ patch_sigmatel.c 2007-09-15 17:50:26.000000000 -0700 @@ -2304,6 +2304,15 @@ stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); + switch (codec->subsystem_id) { + case 0x107b0317: // Gateway MX3412, MT3423 + case 0x107b0318: // Gateway MT3707, ML3109 (ATI SB450) + case 0x10de026c: // Gateway MX3410 + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_POWER_STATE, 0x00); + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); + break; + } + if (spec->gpio_mute) { stac922x_gpio_mute(codec, 0, 0); stac922x_gpio_mute(codec, 1, 0);

2 1

[alsa-devel] 1.0.18: /proc/asound/card0/stream0 no longer existing?
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi there. Can anybody tell me if /proc/asound/card0/stream0 is not existing any longer and what has replaced it? THX a lot Klaus

2 1

Re: [alsa-devel] Trying to figure out what went wrong with my HDMI.
by Clemens Ladisch 17 Dec '08

17 Dec '08

Mark Jenks wrote: > On Tue, Dec 16, 2008 at 7:24 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: > > Use "plughw:0,3" instead of "hw:0,3". > > What is the difference? Plughw automatically converts to a sample format that the device supports. > And why did hw:0,3 work before? Probably your old hardware did support mono playback. > If I want to use it in Myth, would it be ALSA:plughw:0,3 then? Yes. (It is possible that Myth can do resampling and channel rerouting by itself; in this case, hw would also work.) HTH Clemens

1 0

[alsa-devel] alsa-utils: alsactl restore not working after system restart
by Klaus Schulz 16 Dec '08

16 Dec '08

Hi there. I am running Ubuntu 8.10 - and latest ALSA. After restart amixer defaults to default settings. I am not really sure if this is Ubuntu/Debian specific or an Alsa issue. It seems that either a link /var/lib/alsa/asound.state to /etc/asound.state is needed or that /etc/init.d/alsa-utils needs to be updated within the restore section accordingly. With the link workaround I managed to get it working. Cheers Klaus

3 2

[alsa-devel] [PATCH] 1/2 ALSA: ca0106 support 44100Hz playback to spdif
by Ben Stanley 16 Dec '08

16 Dec '08

>From 241939bdf80de14e204d6fcb43d850d33bdeda20 Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Tue, 16 Dec 2008 23:47:36 +1100 Subject: [PATCH] ALSA: ca0106 support 44100Hz playback to spdif This patch provides support for playback of 44100Hz sampled material in 16 and 32 bit sample depths through the digital spdif/iec958 connection/s available on the ca0106 card. I have re-worked the patch to address comments from a previous review by Takashi [1]. In particular, I have incorporated spin-locking to address a race condition that Takashi was concerned about. This code has been running on my MythTV box for the last fortnight without apparent ills. I have re-based the code a bit for submission. This patch is against alsa-kmirror. [1] http://article.gmane.org/gmane.linux.alsa.devel/55825/match=takashi+ca0106+… Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106.h | 13 ++- pci/ca0106/ca0106_main.c | 402 ++++++++++++++++++++++++++++++++++++++------- 2 files changed, 351 insertions(+), 64 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 3faccb6..2ffec74 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.22 + * Version: 0.0.23 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -49,6 +49,8 @@ * Implement support for Line-in capture on SB Live 24bit. * 0.0.22 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.23 + * Add support for playback sampling rate and format constraints. * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -644,6 +646,8 @@ #include "ca_midi.h" +#define DRVNAME "snd-ca0106" + struct snd_ca0106; struct snd_ca0106_channel { @@ -659,6 +663,7 @@ struct snd_ca0106_pcm { struct snd_pcm_substream *substream; int channel_id; unsigned short running; + unsigned short hw_reserved; }; struct snd_ca0106_details { @@ -688,6 +693,7 @@ struct snd_ca0106 { unsigned short model; /* subsystem id */ spinlock_t emu_lock; + spinlock_t pcm_lock; struct snd_ac97 *ac97; struct snd_pcm *pcm; @@ -707,6 +713,11 @@ struct snd_ca0106 { struct snd_ca_midi midi2; u16 spi_dac_reg[16]; + + unsigned char count_pb_44100_chan; + unsigned char count_pb_non_44100_chan; + unsigned char count_pb_S16_chan; + unsigned char count_pb_S32_chan; }; int snd_ca0106_mixer(struct snd_ca0106 *emu); diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 6ac1936..bc78312 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.25 + * Version: 0.0.26 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -83,9 +83,14 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * 0.0.25 * Powerdown SPI DAC channels when not in use + * 0.0.26 Ben Stanley + * Added support for output at 44100Hz rate (SPDIF only). + * Implemented constraints system for output rate and format. * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. + * Some programs fail to produce sound output (tested on SPDIF). See + * http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 * -- * * TODO: @@ -145,6 +150,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/ac97_codec.h> #include <sound/info.h> @@ -285,9 +291,9 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000), - .rate_min = 48000, + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, .rate_max = 192000, .channels_min = 2, //1, .channels_max = 2, //6, @@ -319,6 +325,110 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .fifo_size = 0, }; +static unsigned int all_spdif_playback_rates[] = + {44100, 48000, 96000, 192000}; + +static int hw_rule_playback_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int mask; + unsigned long flags; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + spin_lock_irqsave(&chip->pcm_lock, flags); + if (chip->spdif_enable) { + /* Compute the mask applied to all_spdif_playback_rates */ + if (chip->count_pb_44100_chan) + mask = 0x1; + else if (chip->count_pb_non_44100_chan) + mask = 0xE; + else + mask = 0xF; + } else { + /* 44100Hz is not supported for DAC. + The DAC cannot be clocked at 44100, only 48000 and 96000 + from the CA0106 chip. The SPDIF can be clocked at 44100. + It is a hardware limitation. */ + mask = 0xE; + } + snd_printdd("snd_hw_rule_playback_rate: any_44100=%d, " + "any_non_44100=%d, mask=0x%X, spdif=%d\n", + chip->count_pb_44100_chan, + chip->count_pb_non_44100_chan, + mask, chip->spdif_enable); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return snd_interval_list(hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE), + ARRAY_SIZE(all_spdif_playback_rates), + all_spdif_playback_rates, mask); +} + +static int hw_rule_playback_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + struct snd_mask fmt; + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + int result; + unsigned long flags; + if (snd_BUG_ON(!chip)) + return -EINVAL; + snd_mask_none(&fmt); + + spin_lock_irqsave(&chip->pcm_lock, flags); + if (chip->count_pb_S16_chan) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + else if (chip->count_pb_S32_chan) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + else { + /* No format yet chosen, so both formats are available. */ + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + } + result = snd_mask_refine(f, &fmt); + snd_printdd("snd_hw_rule_playback_format: any_S16=%d, any_S32=%d, " + "refined_fmt=0x%X, avail_fmt=0x%X, changed=%d\n", + chip->count_pb_S16_chan, + chip->count_pb_S32_chan, f->bits[0], fmt.bits[0], + result); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return result; +} + +void snd_ca0106_rebuild_playback_channel_counters(struct snd_ca0106 *chip, + int skip_channel) +{ + /* Re-build the counters of rate=44100 and the formats + for use in constraints processing. Does not process the + current_channel (this must be dealt with in the calling context). + To process all channels, set skip_channel=-1. + This must be called with chip->pcm_lock held. */ + + int chi; + struct snd_ca0106_channel *pchannel; + struct snd_pcm_runtime *runtimei; + + chip->count_pb_44100_chan = chip->count_pb_non_44100_chan = 0; + chip->count_pb_S16_chan = chip->count_pb_S32_chan = 0; + for (chi = 0; chi < 4; ++chi) { + if (chi == skip_channel) + continue; + pchannel = &(chip->playback_channels[chi]); + if (!pchannel->use || !pchannel->epcm + || !pchannel->epcm->hw_reserved) + continue; + runtimei = pchannel->epcm->substream->runtime; + chip->count_pb_44100_chan += runtimei->rate == 44100; + chip->count_pb_non_44100_chan += runtimei->rate != 44100; + chip->count_pb_S16_chan += + runtimei->format == SNDRV_PCM_FORMAT_S16_LE; + chip->count_pb_S32_chan += + runtimei->format == SNDRV_PCM_FORMAT_S32_LE; + } +} + unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, unsigned int reg, unsigned int chn) @@ -393,7 +503,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, int status; int retry; if ((reg > 0x7f) || (value > 0x1ff)) { - snd_printk(KERN_ERR "i2c_write: invalid values.\n"); + snd_printk(KERN_ERR DRVNAME "i2c_write: invalid values.\n"); return -EINVAL; } @@ -463,7 +573,20 @@ static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime) { + struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106 *chip = epcm->emu; + /* FIXME how to tell which case to use? */ + /* struct snd_ca0106_channel *channel = + &(chip->playback_channels[epcm->channel_id]); */ + /* struct snd_ca0106_channel *channel = + &(chip->capture_channels[epcm->channel_id]); */ + unsigned long flags; + + spin_lock_irqsave(&chip->pcm_lock, flags); kfree(runtime->private_data); + runtime->private_data = 0; + /* *channel = 0; */ + spin_unlock_irqrestore(&chip->pcm_lock, flags); } static const int spi_dacd_reg[] = { @@ -487,19 +610,21 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]); struct snd_ca0106_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); if (epcm == NULL) return -ENOMEM; + spin_lock_irqsave(&chip->pcm_lock, flags); epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_playback_hw; channel->emu = chip; @@ -509,10 +634,24 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) - return err; + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + goto error; + err = snd_pcm_hw_constraint_step(runtime, + 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err < 0) + goto error; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_playback_rate, (void *)chip, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto error; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_playback_format, (void *)chip, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + goto error; snd_pcm_set_sync(substream); if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { @@ -522,9 +661,12 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); if (err < 0) - return err; + goto error; } - return 0; + err = 0; +error: + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return err; } /* close callback */ @@ -532,8 +674,12 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + spin_lock_irqsave(&chip->pcm_lock, flags); + epcm = runtime->private_data; chip->playback_channels[epcm->channel_id].use = 0; + spin_unlock_irqrestore(&chip->pcm_lock, flags); if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { const int reg = spi_dacd_reg[epcm->channel_id]; @@ -574,6 +720,7 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]); struct snd_ca0106_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); @@ -584,10 +731,11 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + + spin_lock_irqsave(&chip->pcm_lock, flags); runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_capture_hw; channel->emu = chip; @@ -596,13 +744,16 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre channel->use = 1; //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; - channel->epcm = epcm; + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; + goto error; //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) - return err; - return 0; + goto error; + err = 0; +error: + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return err; } /* close callback */ @@ -610,9 +761,13 @@ static int snd_ca0106_pcm_close_capture(struct snd_pcm_substream *substream) { struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + spin_lock_irqsave(&chip->pcm_lock, flags); + epcm = runtime->private_data; chip->capture_channels[epcm->channel_id].use = 0; /* FIXME: maybe zero others */ + spin_unlock_irqrestore(&chip->pcm_lock, flags); return 0; } @@ -647,6 +802,24 @@ static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream /* hw_free callback */ static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream) { + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + + spin_lock_irqsave(&chip->pcm_lock, flags); + runtime = substream->runtime; + epcm = runtime->private_data; + epcm->hw_reserved = 0; + + snd_ca0106_rebuild_playback_channel_counters(chip, epcm->channel_id); + + snd_printdd(KERN_DEBUG DRVNAME ": free_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + chip->count_pb_44100_chan, chip->count_pb_non_44100_chan, + emu->spdif_enable); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return snd_pcm_lib_free_pages(substream); } @@ -668,53 +841,149 @@ static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream) static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; - int channel = epcm->channel_id; - u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + struct snd_pcm_runtime *runtime = substream->runtime, *runtimei; + struct snd_ca0106_pcm *epcm; + struct snd_ca0106_channel *pchannel; + int channel, chi; + unsigned char count_pb_44100_chan, count_pb_non_44100_chan; + unsigned char count_pb_S16_chan, count_pb_S32_chan; + u32 *table_base; u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; u32 hcfg_set = 0x00000000; u32 hcfg; - u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_mask = 0xFF0000; u32 reg40_set = 0; u32 reg40; - /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ - u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_mask; + u32 reg71_shift; u32 reg71_set = 0; u32 reg71; int i; + unsigned long flags; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + /* FIXME CLEAN UP IF spdif_enable IS CHANGED WHILE CHANNELS ARE OPENED + * OR PREVENT THIS FROM HAPPENING. */ + if (emu->spdif_enable) + reg71_shift = 24; /* SPDIF Output Rate */ + else + reg71_shift = 16; /* I2S Output Rate */ + reg71_mask = 0x3 << reg71_shift; + + /*printk(KERN_DEBUG DRVNAME ": prepare_playback: " + "channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld,period_size=%ld, periods=%u, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1));*/ + /*printk("dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr,runtime->dma_area, table_base);*/ + /*printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr,emu->buffer.area, emu->buffer.bytes);*/ + /* We are forced to build the settings for all the channels. */ + spin_lock_irqsave(&emu->pcm_lock, flags); + epcm = runtime->private_data; + channel = epcm->channel_id; + + snd_ca0106_rebuild_playback_channel_counters(emu, channel); + + /* Determine the effects of the new settings for validation. + Note that nothing is recorded permanently + until the new settings are validated. */ + count_pb_44100_chan = + emu->count_pb_44100_chan + runtime->rate == 44100; + count_pb_non_44100_chan = + emu->count_pb_non_44100_chan + runtime->rate != 44100; + count_pb_S16_chan = emu->count_pb_S16_chan + + runtime->format == SNDRV_PCM_FORMAT_S16_LE; + count_pb_S32_chan = emu->count_pb_S32_chan + + runtime->format == SNDRV_PCM_FORMAT_S32_LE; + /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ - switch (runtime->rate) { - case 44100: - reg40_set = 0x10000 << (channel<<1); - reg71_set = 0x01010000; - break; - case 48000: - reg40_set = 0; - reg71_set = 0; - break; - case 96000: - reg40_set = 0x20000 << (channel<<1); - reg71_set = 0x02020000; - break; - case 192000: - reg40_set = 0x30000 << (channel<<1); - reg71_set = 0x03030000; - break; - default: - reg40_set = 0; - reg71_set = 0; - break; + for (chi = 0; chi < 4; ++chi) { + pchannel = &(emu->playback_channels[chi]); + if (!pchannel->use || !pchannel->epcm) + continue; + if (!pchannel->epcm->hw_reserved && chi != channel) + continue; + runtimei = pchannel->epcm->substream->runtime; + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtimei->rate) { + case 44100: + /* 44100Hz is not supported for DAC. The DAC cannot be + clocked at 44100, only 48000 and 96000 from the CA0106 + chip. The SPDIF can be clocked at 44100. + It is a hardware limitation. */ + if (emu->spdif_enable) { + /* When using 44100, *all* channels + must be set to that rate. */ + reg40_set |= 0x550000; + reg71_set |= 0x1 << reg71_shift; + break; + } else { + printk(KERN_ERR DRVNAME + "prepare_playback: " + "44100Hz is invalid for DAC.\n"); + } + case 48000: + /* reg40_set &= !(0x1 << (chi<<1)); */ + /* reg71_set &= !(0x1 << reg71_shift); */ + break; + case 96000: + reg40_set |= 0x20000 << (chi<<1); + reg71_set |= 0x2 << reg71_shift; + break; + case 192000: + reg40_set |= 0x30000 << (chi<<1); + reg71_set |= 0x3 << reg71_shift; + break; + default: + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + ": prepare_playback: " + "Bad sampling frequency %dHz.\n", + runtime->rate); + return -EINVAL; + } } - /* Format is a global setting */ - /* FIXME: Only let the first channel accessed set this. */ + if (count_pb_44100_chan && count_pb_non_44100_chan) { + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + "prepare_playback: requested sampling rate %dHz" + " conflicts with other selected sampling rates.\n", + runtime->rate); + return -EINVAL; + } + if (count_pb_S16_chan && count_pb_S32_chan) { + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + "prepare_playback: requested sample format %d" + " conflicts with other selected sample formats.\n", + runtime->format); + return -EINVAL; + } + /* Requested sample rate and format are now validated. + Commit the change. */ + emu->count_pb_44100_chan = count_pb_44100_chan; + emu->count_pb_non_44100_chan = count_pb_non_44100_chan; + emu->count_pb_S16_chan = count_pb_S16_chan; + emu->count_pb_S32_chan = count_pb_S32_chan; + epcm->hw_reserved = 1; + + snd_printdd(KERN_DEBUG DRVNAME ": prepare_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + emu->count_pb_44100_chan, emu->count_pb_non_44100_chan, + emu->spdif_enable); + spin_unlock_irqrestore(&emu->pcm_lock, flags); + + /* Format is a global setting. */ + /* Only the first channel accessed can set this + (enforced by constraints). */ switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: hcfg_set = 0; @@ -736,10 +1005,13 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) reg71 = (reg71 & ~reg71_mask) | reg71_set; snd_ca0106_ptr_write(emu, 0x71, 0, reg71); - /* FIXME: Check emu->buffer.size before actually writing to it. */ - for(i=0; i < runtime->periods; i++) { - table_base[i*2] = runtime->dma_addr + (i * period_size_bytes); - table_base[i*2+1] = period_size_bytes << 16; + table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + if (!snd_BUG_ON(8*16*channel+runtime->periods*2 > emu->buffer.bytes)) { + for (i = 0; i < runtime->periods; i++) { + table_base[i*2] = runtime->dma_addr + + (i * period_size_bytes); + table_base[i*2+1] = period_size_bytes << 16; + } } snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel)); @@ -1335,6 +1607,7 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, chip->irq = -1; spin_lock_init(&chip->emu_lock); + spin_lock_init(&chip->pcm_lock); chip->port = pci_resource_start(pci, 0); if ((chip->res_port = request_region(chip->port, 0x20, @@ -1351,8 +1624,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, return -EBUSY; } chip->irq = pci->irq; - - /* This stores the periods table. */ + + /* This stores the periods table. FIXME Only the first 512 bytes + appear to be used (for playback buffers) */ if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { snd_ca0106_free(chip); return -ENOMEM; @@ -1363,8 +1637,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - pci->revision, chip->serial); + printk(KERN_INFO DRVNAME ": Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); @@ -1378,7 +1652,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, } chip->details = c; if (subsystem[dev]) { - printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", + printk(KERN_INFO DRVNAME + ": Sound card name=%s, subsystem=0x%x. " + "Forced to subsystem=0x%x\n", c->name, chip->serial, subsystem[dev]); } -- 1.5.4.3

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[alsa-devel] [PATCH] ALSA: ca0106 MSI K8N Diamond MB spi_dac 2->1
by Ben Stanley 16 Dec '08

16 Dec '08

>From fe2e6692ea6fcdf93d724b8aa48e7433994e011c Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Mon, 15 Dec 2008 23:38:12 +1100 Subject: [PATCH] ALSA: ca0106 MSI K8N Diamond MB spi_dac 2->1 This patch removes an inconsistency that became apparent when I documented the fields of snd_ca0106_details. spi_dac is always used in a 'boolean' sense, so this cleanup should make no difference. Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106_main.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 88fbf28..6ac1936 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -254,7 +254,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB", .gpio_type = 2, .i2c_adc = 1, - .spi_dac = 2 } , + .spi_dac = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". -- 1.5.4.3

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[alsa-devel] [PATCH] ALSA: ca0106 Add comments to snd_ca0106_details struct
by Ben Stanley 16 Dec '08

16 Dec '08

>From dada1c7fe866b860f96d56315648cd5704675aa1 Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Fri, 12 Dec 2008 09:47:13 +1100 Subject: [PATCH] ALSA: ca0106 Add comments to snd_ca0106_details struct Takashi wrote an email [1] explaining the fields of snd_ca0106_details, so I captured the information into the ca0106.h header file. [1] http://article.gmane.org/gmane.linux.alsa.devel/56783/match=takashi+gpio_ty… Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106.h | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 74175fc..3faccb6 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -664,10 +664,14 @@ struct snd_ca0106_pcm { struct snd_ca0106_details { u32 serial; char * name; - int ac97; - int gpio_type; - int i2c_adc; - int spi_dac; + int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in. + ac97 = 1 -> Default to AC97 in. */ + int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in + gpio_type = 2 -> shared side-out/line-in. */ + int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume + controls, phone, mic, line-in and aux. */ + int spi_dac; /* spi_dac=1 adds the mute switch for each analog + output, front, rear, etc. */ }; // definition of the chip-specific record -- 1.5.4.3

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[alsa-devel] Trying to figure out what went wrong with my HDMI.
by Mark Jenks 16 Dec '08

16 Dec '08

I have a M3N78 Pro motherboard, and am running alsa 1.0.18a on it. I had HDMI audio working on it, but for some reason it has stopped. I really didn't change too much about the computer, other then take out a hvr-1800, and put in a hvr-1600. I did try to go up to .27 kernel, but had other problems and went back to .26. Recompiled alsa 1.0.18a and even tried multiple versions of the nvidia drivers. (173, 177).. And now I have just tried the snapshot with the same results. I believe it was recognised as a ALC883 before, and not is is a ALC1200, but I'm not completely sure about it. Help! :) -Mark aplay -l shows: card 0: NVidia [HDA NVidia], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So, everything looks good there. If I try and do an aplay, I get. #aplay -D hw:0,3 /usr/share/sounds/alsa/Noise.wav Playing WAVE '/usr/share/sounds/alsa/Noise.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:966: Channels count non available I noticed that I am missing hdmi in my aplay -L. When it worked, it was there. # aplay -L front:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default

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[alsa-devel] ASoC: help needed for capture DAPM (twl4030)
by Peter Ujfalusi 16 Dec '08

16 Dec '08

Hello, I need some help/advice on how to implement the DAPM routings for twl4030 capture path. There are at least two issues which I'm wondering at the moment. [1] Mux for the analog paths As with other parts, this is implemented as a bitfield, where only one bit can be enabled at the time (0x0 being no routed inputs). ADCL <-|- Main Mix (0x1) |- Headset mic (0x2) |- AUXL, LineL (0x4) |- Carkit mic (0x8) Invalid values: 0x3, 0x5, 0x6, 0x7 ADCR <-|- Sub mic (0x1) |- AUXR, LineR (0x4) Invalid values: 0x2, 0x3 Since the playback path also has similar 'mux as not enum but bitfield' types, I'm thinking of creating a new type for soc based on the soc_enum. This new type would store the valid enum texts and the corresponding value in the register (with mask). This could be used in very flexible way. It can be also be used instead of the normal enum types (altrough it does not make sense). But this can wait for now. [2] ADC versus digital microphone TWL has two ADCs (ADCL, ADCR), the routing looks something like this, when the ADCs are in use: InputL mux -> AmpliL -> ADCL |-> ATXL1_PGA -> Mic channel1 (I2S, TDM) |-> ATXL2_PGA -> Mic channel2 (TDM) InputR mux -> AmpliR -> ADCR |-> ATXR1_PGA -> Mic channel3 (I2S, TDM) |-> ATXR2_PGA -> Mic channel4 (TDM) When the digital microphones are in use: Digimic0 Left -> ATXL1_PGA -> Mic channel1 (I2S, TDM) Digimic0 Right -> ATXR1_PGA -> Mic channel3 (I2S, TDM) Digimic1 Left -> ATXL2_PGA -> Mic channel2 (TDM) Digimic1 Right -> ATXR2_PGA -> Mic channel4 (TDM) There is a single point of selection for Mic channel1,3 and for channel2,4. So ATXL1_PGA and ATXR1_PGA receives the data either from the ADCs or from the digimic0, ATXL2_PGA and ATXR2_PGA receives the data either from the ADCs or from the digimic1. As you can see, if the digital mic is selected the ADC can be powered off, since they are not needed. How to map this with DAPM? Is it mandatory to have the SND_SOC_DAPM_INPUT(digimic) to be connected to SND_SOC_DAPM_ADC? I have been thinking to create four 'ghost ADC' for the four channel. Than when building the route the real ADCs can receive switches to handle the power management. Thank you, -- Péter

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[alsa-devel] [PATCH] ALSA: hda - Add automatic model setting for Samsung Q45
by Luke Yelavich 16 Dec '08

16 Dec '08

Have the Samsung Q45 (144d:c510) select ALC262_HIPPO by default Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4afa0c9..3b2a793 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10541,6 +10541,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), -- 1.5.6.3

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[alsa-devel] HDMI problem, more info.
by Mark Jenks 16 Dec '08

16 Dec '08

More info about the M3n78 Pro motherboard that I am having problems with it showing properly in 'aplay -L'. I turned on ./configure -with-debug=verbose ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x6 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x6 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 # cat /proc/asound/card0/codec#3 Codec: Generic 10de NVIDIA MCP78 HDMI Address: 3 Vendor Id: 0x10de0002 Subsystem Id: 0x10de0101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x18560110: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560121: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560122: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560123: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560124: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0c

1 0

[alsa-devel] [PATCH] ALSA: Fix declaration of sound_class
by Mark Brown 15 Dec '08

15 Dec '08

Include sound/core.h in sound_core.c so that sound_class is declared before it is defined, avoiding it looking like it should be static. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/sound_core.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 10ba421..2b302bb 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -10,6 +10,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/err.h> +#include <sound/core.h> #ifdef CONFIG_SOUND_OSS_CORE static int __init init_oss_soundcore(void); -- 1.5.6.5

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[alsa-devel] Xrun stack trace for 1010LT
by Pete cat 15 Dec '08

15 Dec '08

Im trying to get xrun free recording from a pair of 1010LTs (ice1712) and ive resorted to setting xrun_debug to record stack traces on xruns. Im getting xruns every few minutes (between 2 and 15 minute gaps between single xrun events) and every xrun produces identical stack traces like this: [17127.486287] Pid: 775, comm: IRQ-3 Not tainted 2.6.24.1 #2 [17127.486309] [<c02ade37>] snd_pcm_period_elapsed+0x247/0x2f0 [17127.486325] [<c010312e>] __switch_to+0x9e/0x150 [17127.486336] [<c0128674>] finish_task_switch+0x34/0xd0 [17127.486350] [<c02c2119>] snd_ice1712_interrupt+0xa9/0x1a0 [17127.486364] [<c016ab7c>] handle_IRQ_event+0x5c/0x100 [17127.486379] [<c016b1ba>] thread_simple_irq+0x4a/0x90 [17127.486390] [<c016bb09>] do_irqd+0x229/0x290 [17127.486404] [<c016b8e0>] do_irqd+0x0/0x290 [17127.486411] [<c0141ef2>] kthread+0x42/0x70 [17127.486418] [<c0141eb0>] kthread+0x0/0x70 [17127.486425] [<c0105857>] kernel_thread_helper+0x7/0x10 [17127.486437] ======================= Running the rt kernel (Linux ubuntu-studio 2.6.24.1 #2 SMP PREEMPT RT Thu Dec 4 21:30:26 GMT 2008 i686 GNU/Linux) - although the original installation was Ubuntu studio, the kernel is recompiled. The xruns seem to be all on the same card. IRQs are set to priority of 99 with chrt. Software for record is arecord, modified to use a 10 second ring buffer which is written to disk in a separate thread. I was hoping that from the stack trace I could work out what was causing the xruns, but from the above it dosent look like there is anything causing them. However this is the first time I am using stack traces so maybe im missing something. Can anyone point to the fault for me? Thanks, Pete

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[alsa-devel] Intel 82801I (ICH9 Family) HD Audio Controller on a HP dv4 computer
by Jean-Pierre ANDRE 15 Dec '08

15 Dec '08

Hi, I am getting difficulty setting up ALSA for my new HP portable computer. lspci displays the audio chip as : Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) and /proc/asound displays the codec as IDT 92HD71B7X I have uploaded the detailed hardware and configuration parameters to http://pagesperso-orange.fr/b.andre/visits.html/92HD71B7X I am using Fedora 10 x86_64 without pulseaudio and with the latest alsa-driver : alsa-driver-1.0.18a.16.g4012f.139.g6e583.tar.bz2 I have googled for a similar problem, but did not get to an actual fix, the following thread appears to deal with a most similar situation : http://forums.opensuse.org/pre-release-beta/399731-beta-5-no-sound.html After several tries, I have set modprobe.conf as alias snd-card-0 snd-hda-intel options sound slots=snd-hda-intel options snd-hda-intel model=hp-m4 single_cmd=1 enable_msi=1 Results : aplay freezes, with some sound output which is the continual repeat of the initial expected sound (some fraction of a second repeated). However if I boot with pci=noacpi the output is correct (I do not understand the relation of power management to sound, there must be one as I see : HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 when I unload/reload alsa) WITHOUT the pci=noacpi option : - mplayer outputs the sound correctly (does mplayer bypass alsa ?) - xine outputs a chopped sound (some 100ms OK, some 100ms silent, some 100ms OK, etc, with a period which could be 200..500ms) - most players (eg Real player) just freeze. I would say that when some sound buffer gets empty, the driver does not get the information, so it does not fill the buffer again. I have done some digging into the code. I see that this 92HD71B7X chip is supported for HP dv5 and dv7 computers (subsystem ID 103c30f2 and 103C30f4), but mine is a HP dv4 showing a subsystem ID 103c30f7, so it must be somewhat different and not supported yet. Can anybody suggest some fix ? If needed I can apply patches to the code and do some testing so that my configuration gets supported in subsequent versions. Regards Jean-Pierre

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Re: [alsa-devel] alsa 1.0.18 does not work for Intel onboard cards
by Takashi Iwai 15 Dec '08

15 Dec '08

At Mon, 15 Dec 2008 07:55:22 -0200, Paulo Cavalcanti wrote: > > On Mon, Dec 15, 2008 at 5:31 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Sun, 14 Dec 2008 09:29:03 -0200, > Paulo Cavalcanti wrote: > > > > On Sun, Dec 14, 2008 at 7:12 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Sat, 13 Dec 2008 13:52:24 -0200, > > Paulo Cavalcanti wrote: > > > > > > On Sat, Dec 13, 2008 at 1:23 PM, Paulo Cavalcanti < > promac(a)gmail.com> > > wrote: > > > > > > > Hi, > > > > > > > > alsa driver 1.0.18 has been a nightmare for me. > > > > > > > > I had a perfectly working sound for my Intel D975XBX2 when using > > > > alsa driver 1.017. Going to 1.0.18 just muted my front sound. > > > > Well, maybe not everybody needs a front sound output, but I just > > > > stick with the 1.01.17 driver. > > > > > > > > > > > Then, I tried an Intel DG45ID mobo, which does not have > > > any sound at all, with the later alsa driver 1.0.18a (detected as > a > > Generic > > > DG45). > > > > > > Going back to alsa driver 1.0.17, from the stock Fedora 10 kernel > > > (my card is then detected as an IDT 92HD73E1X5), > > > I have an analogic sound working, but Master, Headphones, and PCM > > > channels have to be above 90%, for anyone being able to hear > anything. > > > Therefore, the volume is compressed in the upper 10% of the scale. > > > > > > Furthermore, I have no mic, and no input working (I would like to > feed > > the > > > output > > > of my capture card to the input source of the sound card) . > > > It seems to me that the input (blue jack) is configured as > output, > > > in the 1.0.17 alsa driver. > > > > > > I do not know what else I can try, maybe some alsa snapshot, > > > to fix some of these issues. > > > > Try the snapshot on > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > alsa-driver-snapshot.tar.gz > > > > Thanks, Takashi > > > > but it did not change anything for the Intel DG45ID. > > This is a Genuine Intel mother board. > > > > The sound card is detected as "Generic 8086 Intel G45 Develk", > > but I do not hear anything, even with all mixer channel at the maximum > > volumes. > > > > This is a Fedora 10, with the latest alsa packages, and > > everything seems OK, the applications work, but I do no hear anything. > > > > With the alsa driver from the latest Fedora kernel > > > > Linux version 2.6.27.7-134.fc10.x86_64 > > > > the card is detected as IDT 92HD73E1X5 and, > > > > although I have to put all the mixer channels at the maximum, the sound > > output works fine, even mid emulation. > > > > During the snapshot compilation, I got dozens of this warning: > > > > > ---------------------------------------------------------------------------------- > > /home/roma/redhat/BUILD/alsa-driver/include/config1.h:65:1: warning: > > "CONFIG_SND_AC97_POWER_SAVE_DEFAULT" redefined > > In file included from <command-line>:0: > > ./include/linux/autoconf.h:260:1: warning: this is the location of the > > previous definition > > CC [M] /home/roma/redhat/BUILD/alsa-driver/misc/ac97_bus.o > > In file included from /home/roma/redhat/BUILD/alsa-driver/include/ > config.h:6, > > from /home/roma/redhat/BUILD/alsa-driver/include/ > > adriver.h:25, > > from /home/roma/redhat/BUILD/alsa-driver/misc/ > ac97_bus.c:1: > > > ----------------------------------------------------------------------------------- > > It's no big issue, I guess. > > > This is the output of alsa-info, with the stock driver (1.0.17): > > Run alsa-info with --no-upload option on both working (1.0.17) and > non-working versions, and attach the generated files. > > This is for the Intel D975XBX2, which lost the front sound with 1.0.18: > Soundcard is detected as Sigmatel STAC9274D. Which I/O are you referring to? The headphone output seems unplugged, and thus the outputs are only from other line-outs, as far as I see the codec info. Takashi

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[alsa-devel] : Can't compile alsa-driver-1.0.18a on x86_64 arch
by Mysth-R 15 Dec '08

15 Dec '08

Hi, I am trying to compile the lastest alsa-driver, but I got an error : patching file usbusx2y.c copying file alsa-kernel/usb/usx2y/usbusx2yaudio.c patching file usbusx2yaudio.c Hunk #11 succeeded at 1057 (offset 1 line). copying file alsa-kernel/usb/usx2y/usx2yhwdeppcm.c patching file usx2yhwdeppcm.c make[3]: quittant le répertoire « /home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a/usb/usx2y » make[2]: quittant le répertoire « /home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a/usb » make[1]: quittant le répertoire « /home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a » make -C /lib/modules/2.6.24.7-rt7/build SUBDIRS=/home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a CPP="gcc -E" CC="gcc" modules make[1]: entrant dans le répertoire « /usr/src/linux-2.6.24.7-rt7 » Makefile:510: /usr/src/linux-2.6.24.7-rt7/arch/x86/Makefile: Aucun fichier ou dossier de ce type make[1]: *** Pas de règle pour fabriquer la cible « /usr/src/linux-2.6.24.7-rt7/arch/x86/Makefile ». Arrêt. make[1]: quittant le répertoire « /usr/src/linux-2.6.24.7-rt7 » make: *** [compile] Erreur 2 Here is my ./configure : ./configure --prefix=/usr --with-cards=hda-intel usb-caiaq usb-audio hrtimer rtctimer hpet --with-sequencer=yes --with-card-options=all --with-oss=no--with-kernel=/usr/src/linux-2.6.24.7-rt7/ CFLAGS="-march=nocona -pipe -O2" Thank you for your help ! cheers, Mysth-R -- * *************************************************************************************** * {^_^} Mysth-R {^_^} * <= Aide Auditive => * * http://myspace.com/mysthr * http://myspace.com/aideauditive * http://mysthr.free.fr/Joomla => Site dédié à l'audio sous Fedora/PlanetCCRMA. * ***************************************************************************************

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[alsa-devel] Latest snapshot - reliable source?
by Klaus Schulz 15 Dec '08

15 Dec '08

Hello Takashi. Can I regard your ftp link below as a kind of reliable source where to get the latest snapshot at any time? Can I also expect that it is compatible to latest official driver release? Because than I'd ask people in (worst) case of trouble to give it a try before spamming the mailing list. ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… THX \Klaus

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[alsa-devel] alsa 1.0.18 does not work for Intel onboard cards
by Paulo Cavalcanti 15 Dec '08

15 Dec '08

Hi, alsa driver 1.0.18 has been a nightmare for me. I had a perfectly working sound for my Intel D975XBX2 when using alsa driver 1.017. Going to 1.0.18 just muted my front sound. Well, maybe not everybody needs a front sound output, but I just stick with the 1.01.17 driver. Then, I -- Paulo Roma Cavalcanti LCG - UFRJ

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[alsa-devel] [PATCH] SOC/davinci: switch DPRINTK to pr_debug()
by Alexander Beregalov 14 Dec '08

14 Dec '08

Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/soc/davinci/davinci-pcm.c | 18 ++++++------------ 1 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bc83e1c..74abc9b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -14,6 +14,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> @@ -24,13 +25,6 @@ #include "davinci-pcm.h" -#define DAVINCI_PCM_DEBUG 0 -#if DAVINCI_PCM_DEBUG -#define DPRINTK(x...) printk(KERN_DEBUG x) -#else -#define DPRINTK(x...) -#endif - static struct snd_pcm_hardware davinci_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -78,8 +72,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; - DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x " - "period_size=%x\n", lch, dma_pos, period_size); + pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " + "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -112,7 +106,7 @@ static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; - DPRINTK("lch=%d, status=0x%x\n", lch, ch_status); + pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; @@ -316,8 +310,8 @@ static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_writecombine(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, (void *) buf->addr, size); + pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " + "size=%d\n", (void *) buf->area, (void *) buf->addr, size); if (!buf->area) return -ENOMEM;

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[alsa-devel] Bugs on aspire one A150
by Maxim Levitsky 13 Dec '08

13 Dec '08

I have just bought an Aspire one A150, XP version, as it was the only available here, and installed ubuntu on it. Bugs I discovered so far: ** 1 - embedded controler works in polling mode, due to this: [ 0.708571] ACPI: EC: non-query interrupt received, switching to interrupt mode [ 1.224043] ACPI: EC: missing confirmations, switch off interrupt mode. Maybe this is the reason for the fact that gnome power manager freezes when I unplug the AC, and freezes often when I try to see battery status. (Note: same is seen on my acer aspire 5720) ** 2 - wireless: not to mention the fact that ath5k wasn't installed by default in ubuntu... wireless more or less works, but kernel log is full of backtraces. Was able to connect to my WPA2 access point. Sometimes wireless fails completely, especially after suspend to ram. Advanced features like monitor/injection work, but when I changed the card's mac address it stopped working. I also noticed that if I then start airodump, then wireless works with new mac. ** 3 - internal mic doesn't work. tried model=acer, model=auto. Overall it seems that alsa misprograms O/B realteck codec. I talk about this later. Have same issue on my acer 5720 ** 4 - wireless led doesn't work. ath5k devs, can you fix this? ** 5 - coretemp doesn't show cpu temperature, I have seen somewhere that atom support same thermal diode as core2 and only patch to detect it is needed. Please include such path in 2.6.28 if exists. ** 6 - both card readers are missing from lspci, is this normal? ** 7 - opengl is broken, I tried running neverball in fullscreen, but it shows wrong/corrupted textures (compiz is running, and working well thought) Now I installed 2.6.28-rc3 (actually latest -git) Sadly nothing improved. I only found few regressions: * Wireless now drops connection from my WPA2 access point after few minutes. I changed temporarily its wireless protection to WEP and open mode, and both work fine, maybe this is related to periodic key exchange WPA does? * System doesn't reboot/shutdown, just hangs. Best regards, Maxim Levitsky.

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Re: [alsa-devel] [ath5k-devel] Bugs on aspire one A150
by Maxim Levitsky 13 Dec '08

13 Dec '08

On Sat, 2008-12-13 at 16:08 +0200, Nick Kossifidis wrote: > 2008/12/13 Maxim Levitsky <maximlevitsky(a)gmail.com>: > > > > Hi, > > > > Indeed gpio #3 controls the led. > > I patched ath5k to use it, and led works now fine. > > > > > > Please share your patch with the rest and also post an lspci -vnn for > your card so we can handle this in a generic way This just a one liner hack diff --git a/drivers/net/wireless/ath5k/base.c b/drivers/net/wireless/ath5k/base.c index 9d2c597..a447937 100644 --- a/drivers/net/wireless/ath5k/base.c +++ b/drivers/net/wireless/ath5k/base.c @@ -2595,6 +2595,11 @@ ath5k_init_leds(struct ath5k_softc *sc) sc->led_pin = 1; sc->led_on = 1; /* active high */ } + + __set_bit(ATH_STAT_LEDSOFT, sc->status); + sc->led_pin = 3; + sc->led_on = 1; + if (!test_bit(ATH_STAT_LEDSOFT, sc->status)) goto out; lspci -vnn: 03:00.0 Ethernet controller [0200]: Atheros Communications Inc. AR242x 802.11abg Wireless PCI Express Adapter [168c:001c] (rev 01) Subsystem: Foxconn International, Inc. Device [105b:e008] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 18 Region 0: Memory at 55200000 (64-bit, non-prefetchable) [size=64K] Capabilities: <access denied> Kernel driver in use: ath5k Kernel modules: ath5k Best regards, Maxim Levitsky

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Re: [alsa-devel] [ath5k-devel] Bugs on aspire one A150
by Maxim Levitsky 13 Dec '08

13 Dec '08

On Tue, 2008-11-11 at 00:12 +0200, Nick Kossifidis wrote: > 2008/11/10 Maxim Levitsky <maximlevitsky(a)gmail.com>: > > Bob Copeland wrote: > >> On Fri, Nov 7, 2008 at 10:26 PM, Maxim Levitsky <maximlevitsky(a)gmail.com> wrote: > >>> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by > >>> default in ubuntu... > >>> wireless more or less works, but kernel log is full of backtraces. > >> > >> As you've seen, these should at least be gone now... > >> > >>> Was able to connect to my WPA2 access point. > >>> Sometimes wireless fails completely, especially after suspend to ram. > >> > >> Did you get noise calibration failures in dmesg at this point? Felix posted > >> a patch recently that handles them better. > > Yep, but they seems to be gone in latest -git too. > > Wireless work fine now, it seems like my iwl3945. > > > >> > >>> Advanced features like monitor/injection work, but when I changed the card's > >>> mac address it stopped working. > >>> I also noticed that if I then start airodump, then wireless works with new > >>> mac. > > Why it doesn't accept new mac?, can this be fixed? > > > > > >> > >>> ** 4 - wireless led doesn't work. > >>> ath5k devs, can you fix this? > >> > >> Currently all the LED code we have in there is for setting a particular gpio > >> for particular laptops (and they differ between models). We only have quirks > >> for IBM and HP, none for Acer. Also, the legacy-hal seems to have a different > >> strategy for 2425 chips. I can try to hack up some code for you to test. > >> > > > > I have seen on the web that madwifi did support the led with some gpio settings > > > > sysctl -w dev.wifi0.ledpin=3 > > sysctl -w dev.wifi0.softled=1 > > > > from https://help.ubuntu.com/community/AspireOne > > > > Best regards, > > Maxim Levitsky > > i think that some led settings are stored on EEPROM, let me check it out... Hi, Indeed gpio #3 controls the led. I patched ath5k to use it, and led works now fine. What I dislike is the way led blinks: While nether I like constant rate blinking implemented in windows driver, I don't like the fact that led blinks too fast, sometimes the blink time is so short that led doesn't show full brightness. I think there should be a minimal interval for blinking. Also I don't know what is assumed better, when led is always on, but turns off when active of vise versa? (Due to short intervals, when I set it for first option, it is almost always on, and I almost don't see the interval when it is off) Best regards, Maxim Levitsky

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[alsa-devel] EMU releases 0404/0202 MAC driver stack as open source!!
by Klaus Schulz 12 Dec '08

12 Dec '08

Hi folks. FYI - see below. Does this makes live easier to get the ALSA driver developed. THX Klaus ---Quote--- SCOTTS VALLEY, CA - December 1, 2008 - E-MU Systems today announced that for the first time it will release as open source the Macintosh driver stack for its award winning USB 2.0 audio interface products - the 0202USB, 0404USB, and TrackerPre. The Software Developer Kit will be available on the SourceForge developer's website in early December and can be found at: https://sourceforge.net/projects/zaudiodrivermac/. The software will be released under the GNU Lesser General Public License or LGPL v3, which enables developers to commercially redistribute products derived from this code base. The Software Developer Kit will include the E-MU USB 2.0 driver interface specification and necessary source code for adding new features, enhancing performance, or porting to other Operating Systems. "We are releasing this code to the public to enable programming enthusiasts the ability to experiment with and to extend the capabilities of our products, which we find being used in a variety of applications outside of the MI space proper. I'm convinced that our clever user base will come up with innovative variations not addressed by our mainstream activities," said Michael Lee, marketing manager for E-MU Systems. "I'm excited to see the results generated by the OpenSource community." ---End Quote---

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