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)

2 6

[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

2 3

[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; }

2 4

[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 */

3 2

[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

2 1

[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 */

2 2