April 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ALSA driver for Native Instruments sound hardware
by Daniel Mack 23 Mar '08

23 Mar '08

Hi, some months after the last posting, we finally managed to finish a new version of the ALSA driver for Native Instruments' external USB sound cards such as "Rig Kontrol 2", "Kore Kontroller", "Audio Kontrol 1" and "Audio 8 DJ". Compared to the last release, the following things were changed: - the streaming logic was completely reworked so that multiple substreams are handled correctly - a Linux input device layer was added to support the knobs, buttons and pedals on "Rig Kontrol 2" and "AudioKontrol 1" - workqueues were entirely removed from the code - most data structures are now allocated within one big struct to avoid multiple, smaller allocations - many more cleanups according to Clemens' reply on my last posting[1] The driver was properly tested with all the mentioned hardware on both record and playback channels. The patchset should apply to the current mecurial alsa-kernel tree. Please let me know what you think. Best regards, Daniel [1] http://article.gmane.org/gmane.linux.alsa.devel/34324/ Signed-off-by: Daniel Mack <daniel(a)caiaq.org> Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de>

4 10

[alsa-devel] ALC882 on Abit AW9D-MAX
by Dan Aloni 10 Jul '07

10 Jul '07

Hello, Yesterday I posted this bug in the bugtracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3013 I have received no reponse yet. I'd like to cooperate on solving this. Thanks, -- Dan Aloni XIV LTD, http://www.xivstorage.com da-x (at) monatomic.org, dan (at) xiv.co.il

2 14

[alsa-devel] alsa on Atmel at91
by Michel Benoit 12 Jun '07

12 Jun '07

Hello I am trying to run alsa on an ARM based target AT91SAM9260 based board. The cpu is connected over i2s (atmel's ssc) to a PCM1725 chip. I modifed the AT73C213 driver available in the latest experimental patches from www.at91.com to behave correctly for the PCM1725 chip. Basically all I have done is remove the spi communication with the AT73C213 that regulates the sound output volume. The PCM chip has no such volume control and I use the i2s interface setup as is to send sample data. I cross compile alsa lib and run the program test/pcm_min which generates as expected random noise on the speaker output. However when I try to run test/pcm I get a divide by zero floating point exception because the library returns a period_size of 0. This despite the fact that the period time is set to 100000 and buffer time is set to 500000. I enabled some debug info in the library and I see the following data dumped to the console. ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (20354 20355) PERIOD_SIZE: 977 PERIOD_BYTES: 3908 PERIODS: (4 5) BUFFER_TIME: (81479 81480) BUFFER_SIZE: 3911 BUFFER_BYTES: 15644 TICK_TIME: 10000 How is it possible that the following code from test/pcm.c err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_period_size(params, &size, &dir); if (err < 0) { printf("Unable to get period size for playback: %s\n", snd_strerror(err)); return err; } returns a period_size of 0 when the debug output shows that the period_size has been calculated to be 977? Can it be that the driver has not initialised the alsa structures properly (the driver file is attached to this email)? Have I mucked something up by removing the spi comm. with the at73c213? Could this be due to a problem in the cross compilation of alsa-lib? I use the following configure command line: CC=arm-linux-gcc ./configure \ --host=arm-linux\ --prefix=$HOME/bsp/rootfs/buildroot/build_arm/root/usr\ --with-softfloat (note: I have also tried it without --with-softfloat) I have tried digging through the alsa lib code but it is not very clear to me where the returned value for period_size comes from. I appreciate any pointers, hints or tips that can help me to solve and/or understand this problem. Thanks in advance! Michel

5 20

[alsa-devel] alsa-lib functions reporting conflicting information with two sound cards
by nick smethurst 14 May '07

14 May '07

Hey guys I am rather confused that I am getting crossed information when calling snd_names_list and snd_pcm_info in order to gather information about pcm devices. I have two sound cards in my development pc: RME Digi 96/8 PCI and an onboard VIA 8237. I'm trying to develop a piece of code that generates a list of objects containing pcm device information, including information on virtual devices. My current idea is to use snd_names_list() to get a full list of devices, and then get information about them by opening each device and querying. The bizzare thing is that snd_names_list() is returning comments that do not agree with the associated device. For example, given the following code: #include <alsa/asoundlib.h> int main() { snd_devname_t * list; snd_config_update(); snd_names_list("pcm", &list); while (list) { if (!strcmp(list->name, "hw:0,0")) { printf("hw:0,0 comment: %s\n", list->comment); break; } list = list->next; } snd_names_list_free(list); snd_pcm_t * pcm; snd_pcm_info_t * pcm_info; snd_pcm_info_malloc(&pcm_info); snd_pcm_open(&pcm, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); snd_pcm_info(pcm, pcm_info); printf("id: %s\n", snd_pcm_info_get_id(pcm_info)); snd_pcm_close(pcm); snd_pcm_info_free(pcm_info); return 0; } The result on my pc is: hw:0,0 comment: Physical Device - VIA 8237 (Duplex) id: Digi96 IEC958 So I'm apparently doing something very wrong here. I would like to ask if there is an easier and more reliable way of getting a list of pcm devices and querying them, and if not, what I am doing wrong here? Regards Nicholas

6 16

[alsa-devel] [PATCH] snd-aoa-i2sbus: use MODULE_DEVICE_TABLE instead of plain MODULE_ALIAS
by Johannes Berg 10 May '07

10 May '07

This patch changes snd-aoa-i2sbus to use MODULE_DEVICE_TABLE instead of a hardcoded MODULE_ALIAS. Thanks to Sylvain Munaut for pointing this out. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- Please apply to the alsa tree. sound/aoa/soundbus/i2sbus/i2sbus-core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) --- wireless-dev.orig/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:08:25.903457795 +0200 +++ wireless-dev/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:09:21.253457795 +0200 @@ -23,9 +23,6 @@ MODULE_LICENSE("GPL"); MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); MODULE_DESCRIPTION("Apple Soundbus: I2S support"); -/* for auto-loading, declare that we handle this weird - * string that macio puts into the relevant device */ -MODULE_ALIAS("of:Ni2sTi2sC"); static int force; module_param(force, int, 0444); @@ -37,6 +34,8 @@ static struct of_device_id i2sbus_match[ { } }; +MODULE_DEVICE_TABLE(of, i2sbus_match); + static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, struct dbdma_command_mem *r, int numcmds)

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[alsa-devel] Fwd: [Alsa-devel] Read the input of a soundcard with ALSA
by Nyx 07 May '07

07 May '07

Hi, I have some problems to read the input of my soundcard using Alsa and I don't really understand how to have access to the input of my soundcard. I follow : http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht… # So, in my program, I have created two handles for capture and playback : char *device = "hw:0,0"; snd_output_t *output; snd_input_t *input; snd_pcm_t *phandle, *chandle; # then I connect the phandle to the out err = snd_output_stdio_attach(&output, stdout, 0); if (..) # Do I have to connect the chandle with the input ? # err = snd_input_stdio_attach(&input, stdin, 0); # I open the access to PCM err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, 0) if (..) err = snd_pcm_open(&chandle, pdevice, SND_PCM_STREAM_CAPTURE, 0) if(..) # Hardware and software Parameters ## Hardware Parameters snd_pcm_hw_params; int err; err = snd_pcm_hw_params_any(phandle, params); if(..); err = snd_pcm_hw_params_set_access(phandle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if(..); err = snd_pcm_hw_params_set_format(phandle, params,SND_PCM_FORMAT_S16); if(..); err = snd_pcm_hw_params_set_channels(phandle, params, 1); if(..); err = snd_pcm_hw_params_set_rate_near(phandle, params, 44100, 0); if(..); err = snd_pcm_hw_params_set_buffer_time_near(phandle, params, 500000, &dir); if(..); err = snd_pcm_hw_params_set_time_near(phandle, params, 100000, &dir), if(..); err = snd_pcm_hw_params(phandle, params); ##Software parameters err = snd_pcm_sw_params(phandle, params); if(..); err = snd_pcm_sw_params_set_start_threshold(phandle, swparams, 0x7fffffff); if(..); err = snd_pcm_sw_params_set_avail_min(phandle, swparams, 4); # same configuration for phandle # Link output to input and start err = snd_pcm_link(chandle, phandle); if(..); err = snd_pcm_start(chandle); if(..); #Then I want to read my input frames_in = 0; in_max = 0; latency = 28; buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2); while (1) { r = readbuf(chandle, buffer, latency, &frames_in, &in_max)); if(..); } # description of the function long readbuf(snd_pcm_t *handle, char *buf, long len, size_t *frames, size_t *max) { long r; do { r = snd_pcm_readi(handle, buf, len); } while (r == -EAGAIN); if (r > 0) { *frames += r; if ((long)*max < r) *max = r; } printf ("r = %s, len %li and buf[%d] %f \r", snd_strerror(r), len,i, buf[i]); return r; } So I compile it and I run it, and I have $ read = Broken Pipe, len = 28 and buf[..] = 0.000 Anyone can help me to read the input of my sound card ? Thank you -- K PS : if I sent this message twice, excuse-me. I haven't confirm my subscription the first time

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[alsa-devel] [PATCH] hda_codec: restore control element values during resume
by Daniel Drake 06 May '07

06 May '07

After resume from suspend-to-RAM, there is no sound output from my Intel HDA hardware: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) Found in a Dell Inspiron 640m, SigmaTel STAC9200 chip. After resuming, sound immediately returns when you adjust the volume one notch in either direction. I reported this on the ALSA bug tracker (#0002989). Further investigation shows that the AC_VERB_SET_AMP_GAIN_MUTE verb is not being executed during resume, which seems strange. Does other hardware really store this value during a suspend/resume cycle? Or maybe the volume is generally controlled through other means on other setups? I'm not entirely sure what the difference is between kcontrol and kcontrol_new structures, but I note that there is already some code to restore the values for kcontrol_new elements. I solved the issue with the following patch. Is it correct? Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux-2.6/sound/pci/hda/hda_codec.c =================================================================== --- linux-2.6.orig/sound/pci/hda/hda_codec.c +++ linux-2.6/sound/pci/hda/hda_codec.c @@ -2248,15 +2248,39 @@ EXPORT_SYMBOL(snd_hda_suspend); int snd_hda_resume(struct hda_bus *bus) { struct list_head *p; + struct snd_ctl_elem_value *val = kmalloc(sizeof(*val), GFP_KERNEL); + + if (!val) + return -ENOMEM; list_for_each(p, &bus->codec_list) { struct hda_codec *codec = list_entry(p, struct hda_codec, list); + struct snd_kcontrol *kctl; + + codec->in_resume = 1; + hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + + + list_for_each_entry(kctl, &bus->card->controls, list) { + memset(val, 0, sizeof(*val)); + val->id = kctl->id; + /* Assume that get callback reads only from cache, + * not accessing the real hardware + */ + if (snd_ctl_elem_read(bus->card, val) < 0) + continue; + snd_ctl_elem_write(bus->card, NULL, val); + } + if (codec->patch_ops.resume) codec->patch_ops.resume(codec); + + codec->in_resume = 0; } + kfree(val); return 0; }

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[alsa-devel] [PATCH] snd-usb-audio: explicitly match Logitech QuickCam
by Daniel Drake 02 May '07

02 May '07

Commit 93c8bf45e083b89dffe3a708363c15c1b220c723 modified the USB device matching behaviour to ignore interface class matches if the device class is vendor-specific. This patch adds an explicit ID match for the Logitech QuickCam Messenger, which has a vendor specific device class (but standards-compliant audio interfaces). This fixes a 2.6.20 regression where the audio component of this device was no longer usable. http://bugs.gentoo.org/show_bug.cgi?id=175715 https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/93822 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3040 Based on a patch from sergiom Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/usb/usbquirks.h =================================================================== --- linux.orig/sound/usb/usbquirks.h +++ linux/sound/usb/usbquirks.h @@ -40,6 +40,21 @@ .bInterfaceClass = USB_CLASS_VENDOR_SPEC /* + * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface + * class matches do not take effect without an explicit ID match. + */ + +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08f0, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, + +/* * Yamaha devices */

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[alsa-devel] ALC260 patch for Packard Bell Easy Note v7900
by Andrea Gozzi 02 May '07

02 May '07

Please see if you can accept this patch that fixes sound/headphones/mic on Packard Bell EN v7900 notebook (ALC260 chipset). The code is not mine, I diff'ed using pshou's (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2189) but since it works perfectly it would be nice to see it in the next alsa release. Andrea

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[alsa-devel] [ALSA][hda-codec] fix-connexant-headphone-mute
by Andrei Popa 02 May '07

02 May '07

Hi, The sound works ok with the built-in speakers on my Fujitsu laptop(amilo pro v3205). But if I plug in headphones in the audio out jack there is no sound in the headphones. With kernel 2.6.20 it worked ok. With 2.6.21 it doesn't. This BUG is reported in ALSA bugtracking system with ID 0003039. Patch that fixes this bug: diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 46e93c6..5edd330 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -636,7 +636,7 @@ static void cxt5045_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0, bits); } /* unsolicited event for HP jack sensing */ @@ -988,10 +988,10 @@ static void cxt5047_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0, bits); /* Mute/Unmute PCM 2 for good measure - some systems need this */ snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0, bits); } /* toggle input of built-in and mic jack appropriately */

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Re: [alsa-devel] Boot failure with 2.6.21 with snd-maestro3
by Giridhar Pemmasani 02 May '07

02 May '07

--- Adrian Bunk <bunk(a)stusta.de> wrote: > On Sun, Apr 29, 2007 at 09:55:18AM -0700, Giridhar Pemmasani wrote: > > When snd-maestro3 (ESS Maestro 3 driver) is enabled, kernel 2.6.21 fails > to > > boot - it seems to stop at various points depending on what other modules > are > > enabled. Disabling snd-maestro3 fixes all issues. Moreover, with > everything > > else being same, using maestro3.c from 2.6.20 kernel and using it in > 2.6.21 > > kernel also works - i.e., system boots fine. I will try to isolate the > issue > > and follow up. > > There were only the two attached patches between 2.6.20 and 2.6.21 to > this driver: > - maestro3: add request_firmware() > - maestro3 - Use quirk list helper function > > 2.6.21 + 2.6.20 maestro3.c works for you. > > Does 2.6.21 + 2.6.20 maestro3.c + patch-maestro3-1 work? 2.6.21 with 2.6.20's maestro3.c with patch-maestro3-1 exhibited the same issue. However, I identified the cause: maestro3 driver in 2.6.21 (even just patch-maestro3-1) tries to load firmware file from user space. But each time request_firmware is called it takes about 1 minute to timeout (because I don't have the firmware installed?). If I let the system take its time, even with maestro3.c from 2.6.21, it boots after timing out twice. So after about 2 minutes of no activity, system does boot. Following are trace messages I added to identify the issue: [ 17.004000] input: TPPS/2 IBM TrackPoint as /class/input/input5 [ 70.904000] sound/pci/maestro3.c (snd_m3_create:2763): -2 [ 70.920000] sound/pci/maestro3.c (snd_m3_create:2774): -2 [ 130.940000] sound/pci/maestro3.c (snd_m3_create:2777): -2 [ 130.956000] sound/pci/maestro3.c (snd_m3_create:2788): -2 [ 130.972000] sound/pci/maestro3.c (snd_m3_create:2794): 0 [ 130.992000] sound/pci/maestro3.c (snd_m3_create:2797): 0 [ 131.012000] sound/pci/maestro3.c (snd_m3_create:2802): 0 [ 131.044000] sound/pci/maestro3.c (snd_m3_create:2804): 0 [ 131.580000] sound/pci/maestro3.c (snd_m3_create:2807): 0 [ 131.596000] sound/pci/maestro3.c (snd_m3_create:2810): 0 [ 131.612000] sound/pci/maestro3.c (snd_m3_create:2814): 0 [ 131.628000] sound/pci/maestro3.c (snd_m3_create:2821): 0 [ 132.364000] device-mapper: ioctl: 4.11.0-ioctl (2006-10-12) initialised: dm-d evel(a)redhat.com Giri __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] porter wanted: AMD Geode SCx200 driver OSS to ALSA
by Martin-Éric Racine 02 May '07

02 May '07

Greetings, I work for a company that develops hardware using the AMD Geode chipset. While our current product line is based on the Geode LX model (which is very well supported in ALSA), our legacy products are based on the earlier Geode SC1200 and SC2200, for which no ALSA support exists. However, AMD has OSS driver source code available at: http://www.amd.com/us-en/ConnectivitySolutions/TechnicalResources/0,,50_233… We basically need this ported to an ALSA snd-scx200 module to replace it. (Would now also be a good time to ask for snd-sc5535audio to be renamed snd-sc5535 for consistence with other driver names?) We could provide a sample SC2200-based unit to whoever volunteers for porting AMD's old OSS code to ALSA, Please contact me for details. The resulting code will be directly submitted to the ALSA project for inclusion in the mainstream kernel, to make sure that everyone can benefit from it. Best Regards, -- Martin-Éric Racine http://q-funk.iki.fi

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[alsa-devel] Possible bug in snd_mixer_selem_set_playback_switch_all
by Tim Burrell 30 Apr '07

30 Apr '07

Hi there, I've encountered a situation where if I very quickly, and repeatedly make calls to snd_mixer_selem_set_playback_switch_all I get some strange undefined behavior. The situation is I'm looping over all mixer elements (on all sound cards) to do a system wide mute / unmute. If I make numerous calls to this function it ends up that some mixers only toggle switches on specific channels (right or left), and sometimes the volume on certain (random) channels gets set to some also random (usually 0, but can be anywhere between 0 and max) level. I've been able to remedy the situation by inserting a 10ms delay in between calls to snd_mixer_selem_set_playback_switch_all. Clearly though this is undesirable. I don't have more than one sound card to test on so I can't be sure if it's a problem with all drivers or just mine (hda-intel). I can certainly provide more information if requested, just let me know what I can do. Or if I'm doing something wrong I'd love to be made aware of something I can do to fix this behavior. Thanks, Tim.

1 0

[alsa-devel] List of all Apple Mac's hda-intel pin configurations
by Ivan N. Zlatev 30 Apr '07

30 Apr '07

I have attached all windows drivers' .ini files (taken from latest windows drivers cd of Boot Camp), which include all pin configuration as well as probably some more useful information (GPIO and stuff). I hope this will help you to make a nice update. [Subsystem ID - configuration] 106B0800 - Board1.ini 106B0600 - Board2.ini 106B0700 - Board2.ini 106B0E00 - Board3.ini 106B0F00 - Board3.ini 106B1600 - Board3.ini 106B1700 - Board3.ini 106B0200 - Board3.ini 106B1E00 - Board3.ini 106B0400 - Board4.ini 106B1A00 - Board4.ini 106B0A00 - Board5.ini 106B2200 - Board5.ini 00000100 - Board4.ini Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work? [Solved]
by Cody Jung 30 Apr '07

30 Apr '07

On 4/29/07, Rene Herman <rene.herman(a)gmail.com> wrote: > On 04/29/2007 09:31 AM, Rene Herman wrote: > > > I've attached (to try and avoid linewrap) the lines that you will need in > > /etc/modprobe.conf. > > No I haven't. > > Rene. > > alias snd-card-0 snd-cs4231 > alias sound-slot-0 snd-cs4231 > > options snd-cs4231 index=0 port=0x534 irq=5 dma1=1 dma2=0 > > install snd-cs4231 echo activate >/sys/bus/pnp/devices/01\:01.00/resources &&\ > sleep 1 &&\ > /sbin/modprobe --ignore-install snd-cs4231 > > remove snd-cs4231 /sbin/modprobe -r --ignore-remove snd-cs4231 &&\ > echo disable >/sys/bus/pnp/devices/01\:01.00/resources > > Wahoo! Thank you, thank you! I ended up not going the /etc/modules way, partially because it wasn't working and partially because I didn't know what I was doing. Instead I added the following 4 lines to a file and included it in my init scripts. echo activate > /sys/bus/pnp/devices/01\:01.00/resources sleep 1 modprobe snd modprobe snd-cs4231 port=0x534 irq=5 dma1=1 dma2=0 And it works great! Thanks again for putting up with my incompetence. ~Cody Jung

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Re: [alsa-devel] Boot failure with 2.6.21 with snd-maestro3
by Michal Piotrowski 29 Apr '07

29 Apr '07

Hi, On 29/04/07, Giridhar Pemmasani <pgiri(a)yahoo.com> wrote: > When snd-maestro3 (ESS Maestro 3 driver) is enabled, kernel 2.6.21 fails to > boot - it seems to stop at various points depending on what other modules are > enabled. Disabling snd-maestro3 fixes all issues. Moreover, with everything > else being same, using maestro3.c from 2.6.20 kernel and using it in 2.6.21 > kernel also works - i.e., system boots fine. I will try to isolate the issue > and follow up. Please revert this patches http://www.stardust.webpages.pl/files/patches/alsa_revert/ and test kernel after each revert. Takashi, I'll add this to known regressions for -stable team. > > Giri Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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[alsa-devel] Can't record sound on Aser Aspire 5100. ATI SB450, RelTek ALC 883
by xinyun liu 29 Apr '07

29 Apr '07

hi, I have an Acer Aspire 5100. I can play sound but with the build in mic, I can't record sound. The driver is the latest HG code. http://pastebin.ca/464406 [root@localhost ~]# lspci -nnv -d1002:437b 00:14.2 0403: 1002:437b (rev 01) Subsystem: 1025:009f Flags: bus master, slow devsel, latency 64, IRQ 19 Memory at c0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- [root@localhost ~]# dmesg | fgrep ALSA > /tmp/1 [root@localhost ~]# vi /tmp/1 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/29/2007 04:18 AM, Cody Jung wrote: > Er...that appears to be a .diff file. Do I have to do something with it > for it to be workable, or...? Yes. You'd have to apply the patch (as it's called) to a recent kernel source tree, reconfigure the kernel adding this new VIVO driver to your config and then recompile and reinstall that kernel. Oh well, you posting to alsa-user and not alsa-devel should've been a hint. Rest assured the driver would've worked nicely :-/ There is another thing you can do. What was posted was just a wrapper around cs4231-lib in much the same way that the existing snd-cs4231 module is and that module you _will_ already have installed. That one doesn't do anything automatic though, so you will have to know how to manually enable your card first. First, open a root shell and look through the /sys/bus/pnp/devices directory for your VIVO. Under the "xx:yy.zz" sub-directories there you will find "id" files, and one of them will say "ENS1011". Example: # cat /sys/bus/pnp/devices/01\:01.00/id ENS1011 After having found it in there, activate the card: # echo activate >/sys/bus/pnp/devices/01\:01.00/resources and look at what resources were assigned to it by the PnP layer: # cat /sys/bus/pnp/devices/01\:01.00/resources state = active io 0x330-0x33f io 0x530-0x537 io 0x2100-0x213f irq 5 irq 9 dma 1 dma 0 You can now load the snd-cs4231 module specifying those card resources manually: # modprobe snd-cs4231 port=0x534 irq=5 dma1=1 dma2=0 Yes, 0x534. The second io resource + 4. This should've loaded without problem. Start up alsamixer: # alsamixer and up and unmute the PCM volume. With -> and <- you can walk through the sliders; ^ ups the one you're on, and "m" toggles the mute (the "MM" at the bottom should turn to a highlighted "OO"). PCM is the first slider. 90 seems to be an okay value. Quit alsamixer again with Alt-Q and now you should be able to hear things. If you have a plain "wav" file around somewhere, that makes for the quickest test: # aplay foo.wav No, I'm not going to tell you to plug in your speakers. If anything in the above doesn't work as expected, your system is not as expected and there's isn't much I can help with. If everything does work, you can automate it for the next time. I've attached (to try and avoid linewrap) the lines that you will need in /etc/modprobe.conf. Ubuntu has no doubt found a way to make that dark and muddy again so you might need to add it to an /etc/modprobe/whatever file instead. Please ask a fellow ubuntu victim for advice about translating this generic linux advice to your install. Once you do have that all setup, sound should "Just Work". You probably need to adjust the volume as specified above once only after which ubuntu will save it for the next boot, but if you don't hear anything ater rebooting, please check that first. Rene.

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/28/2007 07:06 AM, Cody Jung wrote: > As it suggests in the subject, I'm both a Linux-newbie and an > ALSA-newbie. I have an old clunker of a machine running Ubuntu Feisty > that I built out of parts left on curbs during spring cleanup, and that's > how I came across my sound card. It's an Ensoniq SoundscapeVivo (Not a > Vivo90, AFAIK). As far as I'm aware, there are no other VIVO's than the VIVO90. > I've gone through install and troubleshooting guides galore, but I can't > seem to get ALSA to recognize my card, and I'm confused and rather tired > of it all. > > Do you think you can help, and if so, what information do you need from > me? I probably can; I have both a Soundscape VIVO90 and a much older Soundscape S-2000. My VIVO90 is an ISA-PnP card and has a Ensoniq MARK5-RO, an OTTOR2C and a Analog Devices AD1845 codec: http://en.wikipedia.org/wiki/Ensoniq_Soundscape_VIVO90 I tested that when I acquired it a while ago and if I remember correctly, it just worked... mmm. come to think of it, it might be the case that I needed to add its PnP ID to the driver. Not sure anymore. Could you confirm whether or not your Soundscape is an ISA-PnP card and if so (that's also a way to find out if you don't know if it is) run "pnpdump" with it installed and look for a PnP-ID for the card in it? Something of the form "ENSxxxx". If your card is one of the older non-PnP soundscapes it's going to be a bit harder; they also need firmware and all that and I haven't tested that card myself yet. (please use the @alsa-project.org list addresses by the way; in the case of alsa-devel, the sourceforge one is gone even). Rene.

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[alsa-devel] Apple iMac pin configs
by Ivan N. Zlatev 28 Apr '07

28 Apr '07

Hello, I have a 20'' iMac machine with a hda-intel (SigmaTel 9221 A1 codec) and I have successfully been using the mac mini pin config, but there is one problem - the microphone doesn't work. The machine has a built-in microphone, built-in speakers, 1 digitial out jack and 1 for headphones. With the mac mini pin config I get 2 mics: "Front Mic" and "Mic" but none of those works. (alsamixer-macmini.gz for info). Today I spent some time on the search for the the correct pin configuration and it should be this [1] according to the windows drivers. This configuration causes amixer.gz, basically 1 "Mic" (which again doesn't record), but also "Mic as Output" and "Line in as output" (I don't have a line in as far as i know). I have also tried the macbook-pro and macbook introduced with 1.0.14rc3, but the mic doesn't work. I have attached * cat /proc/asound/card0/codec#0 * Mac OS X's complete dump of information of the audio driver - imac20-ioreg-audio.gz * amixer with the below config and the one with the macmini pin config Can you possible assist me to get the below pin config in tune, so that everything works? Regards. [1] ========== [HKR\Settings\pin] [HKR\Settings\Pin\0A] CfgDflt = dword: 0x0121E230 Action = hex: 01,0A,81,0C,81,10 [HKR\Settings\Pin\0B] CfgDflt = dword: 0x90A70120 [HKR\Settings\Pin\0C] CfgDflt = dword: 0x9017E110 [HKR\Settings\Pin\0D] CfgDflt = dword: 0x400000FE [HKR\Settings\Pin\0E] CfgDflt = dword: 0x400000FD Action = hex: 01,10,81,0A,81,0C [HKR\Settings\Pin\0F] CfgDflt = dword: 0x0181E02E Action = hex: 09,0F [HKR\Settings\Pin\10] CfgDflt = dword: 0x1145E240 [HKR\Settings\Pin\11] CfgDflt = dword: 0x500000FA [HKR\Settings\Pin\15] CfgDflt = dword: 0x400000FC [HKR\Settings\Pin\1B] CfgDflt = dword: 0x400000FB -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 27 Apr '07

27 Apr '07

Thanks to the help I've gotten on this list my alsa driver is working great. I got a little hung up not realizing that the alsa core basically provides everything you need to implement your own circular buffer if you need to use an intermediate buffer. The only question I still have is that if alsa allocates your DMA buffer for you based on snd_pcm_hardware_t buffer_bytes_max what exactly does snd_pcm_lib_preallocate_pages_for_all allocate buffers for? Does this allocate the buffers that would be passed to the copy callback? Thanks, Tim --- Tim Harvey <tim_harvey(a)yahoo.com> wrote: > Greetings, > > I'm setting out to write an ALSA driver for a very limited device thats > behind > a PLD. The actual codec is a uda1380 and does 16bit MSB stereo > encode/decode. > The way the device is driven the sampling rate is limited to 8KHz and data > transfer to/from the device I have to handle manually. > > I've read over the 'writing an ALSA driver' tutorial and have started the > driver but have run into a few questions: > > - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB > samples does the following snd_pcm_hardware_t look right?: > > static snd_pcm_hardware_t my_playback_hw = { > .info = (SNDRV_PCM_INFO_MMAP | > SNDRV_PCM_INFO_INTERLEAVED | > SNDRV_PCM_INFO_BLOCK_TRANSFER | > SNDRV_PCM_INFO_MMAP_VALID), > .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? > .rates = SNDRV_PCM_RATE_8000, // seems redundant? > .rate_min = 8000, > .rate_max = 8000, > .channels_min = 2, > .channels_max = 2, > .buffer_bytes_max = 32768, > .period_bytes_min = 4096, > .period_bytes_max = 32768, > .periods_min = 1, > .periods_max = 1024, > }; > > - I'm a little confused at how to deal with copying data from/to the device. > > It looks like I would implement the copy callback and copy the passed buffer > to > the device in the device specific manner I need to (playback), but I'm not > clear where to put data that I receive from the device (capture). I notice > that when the hw_params callback is called a dma buffer has been created by > ALSA in the substream->runtime struct. I assume this is created based on hte > buffer_bytes_max field above and I'm to copy data to this buffer? How do I > know where to copy it within the buffer and what data ALSA has already > consumed > from the buffer? > > - I might need to create an intermediate buffer between ALSA's rather large > buffers and my hardware buffer. Is there any driver that I could look at > that > does this? The tutorial talks about the vxpocket driver but that code looks > very foriegn from the rest of the ALSA drivers. > > - I'm not sure I understand what the 'period' is about. Its clear to me > that > I need to call snd_pcm_period_elapsed() periodically from either an IRQ or > other timed callback, but I'm not clear what the hardware pointer is or how > that converts to 'frames' and 'periods'. > > - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA > provides a mechanism for converting data from other formats? > > Thanks for any pointers, > > Tim >

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[alsa-devel] [ASoC] list corruption during AC97 registration
by Manuel Lauss 27 Apr '07

27 Apr '07

Hi Folks, I hit the following oops (on 2.6.21) when ASoC registers an AC97 codec. Happens both on MIPS and SH, for example the mips oops: asoc: AC97 HiFi <-> au1xpsc-ac97-1 mapping ok list_add corruption. prev->next should be next (80488f60), but was 865ffad8. (prev=865ffad8). <snip> Call Trace: [<80271290>] __list_add+0x68/0x94 [<802ac0e4>] bus_attach_device+0x48/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<c00882a8>] snd_soc_register_card+0x1d0/0x2c8 [snd_soc_core] [<c00b3268>] ac97_soc_probe+0x17c/0x1e4 [snd_soc_ac97] [<c0088f48>] soc_probe+0xb0/0x1ac [snd_soc_core] [<802ad410>] really_probe+0x148/0x268 [<802ac1cc>] bus_for_each_drv+0x50/0xa0 [<802ad758>] device_attach+0xe4/0x194 [<802ac0cc>] bus_attach_device+0x30/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<802aff24>] platform_device_add+0x13c/0x198 [<c00b71c8>] exm1200_mobo_ac97_init+0x1c8/0x1fc [snd_soc_exmmb_ac97] [<801478d0>] sys_init_module+0x1448/0x15b8 [<8010bcc0>] stack_done+0x20/0x3c System.map says: 80488e60 D ac97_bus_type As I said, it happens on both MIPS and SH (with machine and PCM drivers written by myself). With CONFIG_DEBUG_LIST disabled the code works and plays sound, but still I'd like to know what causes this list corruption. Thanks for any hints! -- Manuel Lauss

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[alsa-devel] dsnoop issues
by Gordon Miller 27 Apr '07

27 Apr '07

Hi All my snd_pcm_readi calls return too few frames. Also when I switch to stereo, the sound is muted. I cannot change this by adjusting period & buffer sizes. dsnoop is enabled by default. The documentation for snd_pcm_readi says that the only way that the frame count can be short is because of a signal or xrun. Well this clearly isn't the case. I think that direct snoop is the cause of this. Are there any remedies ? because I am roundly SICK of being told that my application must be wrong "because everyone else's works". I have spent the last 3 weeks debugging an app - that always worked properly - because of the faulty dsnoop implementation. It is abundantly clear that dsnoop is far from ready for use, but I am prepared to put some work into this. Whose idea was it to enable dsnoop by default, when it doesn't even work ? I was quite happy with OSS. Changing everything to be alsa entailed a lot of work, and I am beginning to think it was a complete waste of my time. ICE1712, 2.6.20.6 kernel, alsa 1.0.14rc1 Gordon

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[alsa-devel] New bugtracker categories
by Liam Girdwood 27 Apr '07

27 Apr '07

I'd like to request the following new categories for the ALSA bugtracker :- ASoC - platform - s3c24xx ASoC - platform - pxa2xx ASoC - platform - i.MX ASoC - platform - AT91xxxx ASoC - codecs ASoC - core Thanks Liam

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[alsa-devel] [ASoC] Sample format non available
by Markus Korber 27 Apr '07

27 Apr '07

Dear list, I'm currently implementing a ASoC solution for a custom chip and I've written a codec, I2S, PCM, and machine driver. However, when using aplay I get the following error (requested format was: 2): root:~# aplay -M -D hw:0,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:906: Sample format non available However, I've told each piece involved, that it is capable of playing S16_LE. So I don't know what I am missing here? Should I provide more debugging output like RULES_DEBUG output from sound/core/pcm_native.c? ,----[ CODEC DAI ] | #define CS4265_RATES (SNDRV_PCM_RATE_8000_192000) | #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) | | struct snd_soc_codec_dai cs4265_dai = { | .name = "CS4265", | .playback = { | .stream_name = "Playback", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | .capture = { | .stream_name = "Capture", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | [...] | }; `---- ,----[ CPU DAI ] | #define CHIP_I2S_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ | SNDRV_PCM_RATE_48000) | #define CHIP_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) | | struct snd_soc_cpu_dai chip_i2s_dai = { | /* DAI description */ | [...] | /* DAI callbacks */ | [...] | /* DAI capabilities */ | .playback = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | .capture = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | /* ops */ | [...] `---- ,----[ HW DAI ] | static const struct snd_pcm_hardware chip_pcm_hardware = { | .info = SNDRV_PCM_INFO_MMAP | | SNDRV_PCM_INFO_MMAP_VALID | /* For OSS emulation */ | SNDRV_PCM_INFO_BLOCK_TRANSFER | /* For OSS emulation */ | SNDRV_PCM_INFO_NONINTERLEAVED, | .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | | .channels_min = 2, /* Stereo */ | .channels_max = 2, | .period_bytes_min = 32, | .period_bytes_max = 8192, | .periods_min = 2, | .periods_max = 1024, /* PAGE_SIZE/sizeof(chip_dma_desc), */ | .buffer_bytes_max = 32 * 1024, | .fifo_size = 0, /* Still unused in ALSA? */ | }; `---- root:~# uname -a Linux 2.6.21 root:~# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). root:~# aplay --version aplay: version 1.0.14rc2 by Jaroslav Kysela <perex(a)suse.cz> root:~# aplay -M test.wav chip-pcm: Entered chip_pcm_open board_cs4265: Entered board_cs4265_startup Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA lib pcm_plug.c:840:(snd_pcm_plug_hw_refine_cchange) Unable to find an usable client format chip-pcm: Entered chip_pcm_hw_free chip-cs4265: Entered cs4265_shutdown chip-cs4265: Entered cs4265_write board_cs4265: Entered board_cs4265_shutdown chip-pcm: Entered chip_pcm_close ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IEC958_SUBFRAME_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: MU_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: A_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IMA_ADPCM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: GSM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: SPECIAL ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3BE aplay: set_params:878: Broken configuration for this PCM: no configurations available Regards, Markus Korber

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