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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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