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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[alsa-devel] How to play a pcm file
by Flávio Yuiti 26 Apr '07

26 Apr '07

Greetings, I'm a new user of alsa api and I need to play pcm, adpcm and wav files using Alsa api. My code is below. I did a test with pcm file and the sound is playing very fast, the properties of the pcm file are 128kbps, 16 bits little endian, 1 channel, 8 khz, but I don't know how to set correctly the parameters in api. Please help me with this problem. Point me my errors. Thanks, Flavio int Sound::PlayFile(char *pszFileName, ULONG initialPoint, UINT qtdTimesToPlay, BOOL mix) { int dir; int result; int size; unsigned int val; snd_pcm_hw_params_t *params; snd_pcm_uframes_t frames; char* buffer; long tamanhoBuffer; FILE* file; int bytesLidos; // Open PCM device for playback. if((result = snd_pcm_open(&handleSound,"hw:0,0",SND_PCM_STREAM_PLAYBACK,0)) < 0) { fprintf(stderr,"\nUnable to open pcm device: %s",snd_strerror(result)); exit(1); } // Allocate a hardware parameters object. snd_pcm_hw_params_alloca(&hardwareParam); // Fill it in with default values. snd_pcm_hw_params_any(handleSound,hardwareParam); // Set the desired hardware parameters. // Interleaved mode. snd_pcm_hw_params_set_access(handleSound,hardwareParam,SND_PCM_ACCESS_RW_INTERLEAVED); // Signed 16-bit little-endian format. snd_pcm_hw_params_set_format(handleSound,hardwareParam,SND_PCM_FORMAT_S16_LE); // Two channels (stereo), one channel mono. snd_pcm_hw_params_set_channels(handleSound,hardwareParam,1); // Set rate. //snd_pcm_hw_params_set_rate(handleSound,hardwareParam,8000,0); // Set period size to 16384 frames. //snd_pcm_hw_params_set_period_size(handleSound,hardwareParam,16384,0); // Write the parameters to the driver. if((result = snd_pcm_hw_params(handleSound,hardwareParam)) < 0) { fprintf(stderr,"\nUnable to set hw parameters: %s",snd_strerror(result)); exit(1); } tamanhoBuffer = 16384; if((result = snd_pcm_prepare(handleSound)) < 0) { fprintf(stderr,"Cannot prepare audio interface for use: %s\n",snd_strerror(result)); exit(1); } file = fopen(pszFileName,"rb"); buffer = (char *)malloc(tamanhoBuffer); while((bytesLidos = fread(buffer,1,tamanhoBuffer,file)) > 0) { if((result = snd_pcm_writei(handleSound,buffer,tamanhoBuffer)) != tamanhoBuffer) { fprintf(stderr,"Write to audio interface failed: %s\n",snd_strerror(result)); exit(1); } } snd_pcm_drain(handleSound); snd_pcm_close(handleSound); free(buffer); fclose(file); return OK; } __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 2

[alsa-devel] [PATCH] PnP: Protocol for Media Vision Jazz16 sound cards
by Rask Ingemann Lambertsen 26 Apr '07

26 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add code to detect and PnP protocol to manage Media Vision Jazz16 sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20.6. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch depends on the patch "ALSA: Support Media Vision Jazz16 chips" because the patched include/sound/sb.h is needed. drivers/pnp/Kconfig | 12 drivers/pnp/Makefile | 1 drivers/pnp/jazz16pnp.c | 591 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 604 insertions(+) Main changes since the last patch: Request just one DMA channel (1, 3, 5 or 7). Request port 0x388-0x38b for the OPL3 alias to avoid conflicts. Use the recommended resources from the manual for the preferred option. More SB port bases: 0x210, 0x230 and 0x250. diff -rup linux-2.6.20.6-clean/drivers/pnp/Kconfig linux-2.6.20.6-ril/drivers/pnp/Kconfig --- linux-2.6.20.6-clean/drivers/pnp/Kconfig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/drivers/pnp/Kconfig 2007-04-07 20:10:35.000000000 +0200 @@ -37,5 +37,17 @@ source "drivers/pnp/pnpbios/Kconfig" source "drivers/pnp/pnpacpi/Kconfig" +# +# Jazz16 Plug and Play configuration +# +config JAZZ16PNP + bool "Jazz16 sound card Plug and Play support" + depends on PNP && ISA + help + Say Y here if you would like support for automatic configuration + of I/O ports, DMA channels and IRQ lines of Jazz16 sound cards. + + Say Y if you have such a card. Otherwise, say N. + endmenu diff -rup linux-2.6.20.6-clean/drivers/pnp/Makefile linux-2.6.20.6-ril/drivers/pnp/Makefile --- linux-2.6.20.6-clean/drivers/pnp/Makefile 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/drivers/pnp/Makefile 2007-04-07 20:10:35.000000000 +0200 @@ -7,3 +7,4 @@ obj-y := core.o card.o driver.o resourc obj-$(CONFIG_PNPACPI) += pnpacpi/ obj-$(CONFIG_PNPBIOS) += pnpbios/ obj-$(CONFIG_ISAPNP) += isapnp/ +obj-$(CONFIG_JAZZ16PNP) += jazz16pnp.o --- /dev/null 2007-04-11 19:30:17.320748640 +0200 +++ linux-2.6.20.6-ril/drivers/pnp/jazz16pnp.c 2007-04-11 21:41:39.000000000 +0200 @@ -0,0 +1,591 @@ +/* + * Media Vision Jazz16 Plug & Play support. + * + * Copyright (c) 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * This driver manages two devices: + * + * Device 0: The game port. + * Device 1: The SoundBlaster look-alike part. + * + * Sources: + * sound/oss/sb_common.c + * sys/dev/isa/sbdsp.c (OpenBSD) + */ + +#include <linux/types.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/sb.h> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16 Plug & Play support"); +MODULE_LICENSE("GPL"); + +int jazz16pnp_disable = false; /* Disable Jazz16 PnP. */ +module_param_named(disable, jazz16pnp_disable, bool, false); +MODULE_PARM_DESC(disable, "Jazz16 Plug & Play disable switch"); + +int jazz16pnp_verbose = false; /* Verbosity level. */ +module_param_named(verbose, jazz16pnp_verbose, bool, false); +MODULE_PARM_DESC(verbose, "Jazz16 Plug & Play verbosity level"); + +#define PFX "jazz16pnp: " +#define PFX_DEBUG KERN_DEBUG PFX +#define PFX_INFO KERN_INFO PFX +#define PFX_WARN KERN_WARNING PFX + +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_JAZZ16_SET_DMAINTR 0xFB + +extern void *pnp_alloc(long size); +extern void pnp_free_option(struct pnp_option *option); + +static struct pnp_dev *jazz16_gameport_dev = NULL; +static struct pnp_dev *jazz16_sb_dev = NULL; +static struct resource *jazz16pnp_fm_res = NULL; + +static uint jazz16_sb_port = 0, jazz16_sb_irq = 0; +static uint jazz16_sb_dma1 = 0, jazz16_sb_dma2 = 0; +static uint jazz16_mpu401_port = 0, jazz16_mpu401_irq = 0; + +static uint jazz16pnp_count = 0; + +static struct pnp_protocol jazz16pnp_protocol; + +static bool __init jazz16pnp_easy_option(struct pnp_dev *dev, uint pri, + struct pnp_port *port, uint nports, + struct pnp_irq *irq, uint nirqs, + struct pnp_dma *dma, uint ndmas) +{ + struct pnp_option *opt; + struct pnp_port *new_port; + struct pnp_irq *new_irq; + struct pnp_dma *new_dma; + uint i; + + switch (pri) { + case PNP_RES_PRIORITY_PREFERRED: + case PNP_RES_PRIORITY_ACCEPTABLE: + case PNP_RES_PRIORITY_FUNCTIONAL: + opt = pnp_register_dependent_option(dev, pri); + break; + + default: + opt = pnp_register_independent_option(dev); + break; + } + if (!opt) + return false; + + for (i = 0; i < nports; i ++) { + new_port = pnp_alloc(sizeof(*new_port)); + if (!new_port) + goto out_no_mem; + memcpy(new_port, &port[i], sizeof(*new_port)); + pnp_register_port_resource(opt, new_port); + } + for (i = 0; i < nirqs; i ++) { + new_irq = pnp_alloc(sizeof(*new_irq)); + if (!new_irq) + goto out_no_mem; + memcpy(new_irq, &irq[i], sizeof(*new_irq)); + pnp_register_irq_resource(opt, new_irq); + } + for (i = 0; i < ndmas; i ++) { + new_dma = pnp_alloc(sizeof(*new_dma)); + if (!new_dma) + goto out_no_mem; + memcpy(new_dma, &dma[i], sizeof(*new_dma)); + pnp_register_dma_resource(opt, new_dma); + } + return true; + +out_no_mem: + return false; +} + +static struct pnp_port jazz16pnp_game_ports[] __initdata = { + { .min = SB_JAZZ16_CONFIG_PORT, + .max = SB_JAZZ16_CONFIG_PORT, + .size = 1, .align = 0, + .flags = PNP_PORT_FLAG_FIXED, }, +}; + +static int __init jazz16pnp_add_gameport(void) +{ + struct pnp_dev *dev; + struct pnp_id *dev_id; + + if (!(dev = pnp_alloc(sizeof(*dev)))) + goto out_no_dev; + if (!(dev_id = pnp_alloc(sizeof(*dev_id)))) + goto out_no_id; + if (!jazz16pnp_easy_option(dev, PNP_RES_PRIORITY_INVALID, + jazz16pnp_game_ports, 1, + NULL, 0, NULL, 0)) + goto out_no_option; + + dev->protocol = &jazz16pnp_protocol; + dev->capabilities = PNP_READ; + dev->active = 1; + dev->number = 0; + + pnp_init_resource_table(&dev->res); + dev->res.port_resource[0].start = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].flags = IORESOURCE_IO; + + memcpy(dev_id->id, "PNPb02f", PNP_ID_LEN); + pnp_add_id(dev_id, dev); + + jazz16_gameport_dev = dev; + pnp_add_device(dev); + return 0; + +out_no_option: + if (dev->independent) + pnp_free_option(dev->independent); + kfree(dev_id); +out_no_id: + kfree(dev); +out_no_dev: + return -ENOMEM; +}; + +static struct pnp_port jazz16pnp_pref_ports[] __initdata = { + { .min = 0x220, .max = 0x220, .align = 0x00, .size = 0x10 }, + { .min = 0x330, .max = 0x330, .align = 0x00, .size = 0x02 }, + { .min = 0x388, .max = 0x388, .align = 0x00, .size = 0x04 }, +}; + +static struct pnp_port jazz16pnp_ports[] __initdata = { + { .min = 0x210, .max = 0x260, .align = 0x10, .size = 0x10 }, + { .min = 0x300, .max = 0x330, .align = 0x10, .size = 0x02 }, + { .min = 0x388, .max = 0x388, .align = 0x00, .size = 0x04 }, +}; + +static struct pnp_irq jazz16pnp_pref_irqs[] __initdata = { + { .map[0] = (1 << 5), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 9), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_irq jazz16pnp_irqs[] __initdata = { + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_dma jazz16pnp_pref_dmas[] __initdata = { + { .map = (1 << 1), .flags = IORESOURCE_DMA_8BIT, }, +}; + +static struct pnp_dma jazz16pnp_dmas[] __initdata = { + { .map = (1 << 1) | (1 << 3) | (1 << 5) | (1 << 7), + .flags = IORESOURCE_DMA_8AND16BIT, }, +}; + +static int __init jazz16pnp_add_sb(void) +{ + struct pnp_dev *sbdev; + struct pnp_id *sbid; + + if (!(sbdev = pnp_alloc(sizeof(*sbdev)))) + goto out_no_sbdev; + if (!(sbid = pnp_alloc(sizeof(*sbid)))) + goto out_no_sbid; + + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_PREFERRED, + jazz16pnp_pref_ports, 3, + jazz16pnp_pref_irqs, 2, + jazz16pnp_pref_dmas, 1)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 3, + jazz16pnp_irqs, 2, + jazz16pnp_dmas, 1)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 3, + jazz16pnp_irqs, 1, + jazz16pnp_dmas, 1)) + goto out_no_sboption; + memcpy(sbid->id, "PNPb00f", PNP_ID_LEN); + pnp_add_id(sbid, sbdev); + + pnp_init_resource_table(&sbdev->res); + sbdev->capabilities = PNP_READ | PNP_WRITE | PNP_CONFIGURABLE | PNP_DISABLE; + sbdev->protocol = &jazz16pnp_protocol; + sbdev->number = 1; + + jazz16_sb_dev = sbdev; + pnp_add_device(sbdev); + return 0; + +out_no_sboption: + if (sbdev->dependent) + pnp_free_option(sbdev->dependent); + kfree(sbid); +out_no_sbid: + kfree(sbdev); +out_no_sbdev: + return -ENOMEM; +} + +/* Configure the SB and MPU ports. Passing 0 for sbport disables the Jazz16. */ +static void jazz16_port_setup(const uint sbport, const uint mpuport) +{ + u8 config; + + switch (sbport) { + case 0x210: + case 0x220: + case 0x230: + case 0x240: + case 0x250: + case 0x260: + config = sbport & 0x70; + jazz16_sb_port = sbport; + break; + + default: + config = 0; + jazz16_sb_port = 0; + break; + } + + switch (mpuport) { + case 0x300: + case 0x310: + case 0x320: + case 0x330: + config |= (mpuport >> 4) & 0x07; + jazz16_mpu401_port = mpuport; + break; + + default: + if (jazz16_sb_port) + BUG(); + } + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB port = %#x, MPU port = %#x.\n", + jazz16_sb_port, jazz16_mpu401_port); + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(config, SB_JAZZ16_CONFIG_PORT); +} + +static u8 jazz16_irq_config(uint irq) +{ + switch (irq) { + case 3: return 3; + case 5: return 1; + case 7: return 4; + case 9: return 2; + case 10: return 5; + case 15: return 6; + default: return 0; + } +} + +static u8 jazz16_dma_config(uint dma) +{ + switch (dma) { + case 1: return 1; + case 3: return 2; + case 5: return 3; + case 7: return 4; + default: return 0; + } +} + +/* The my_sbdsp_xxx functions were ripped from sound/isa/sb/sb_common.c. */ +#define BUSY_LOOPS 100000 + +static int my_sbdsp_command(unsigned int base, unsigned char val) +{ + int i; + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(base, STATUS)) & 0x80) == 0) { + outb(val, SBP1(base, COMMAND)); + return 1; + } + return 0; +} + +static int my_sbdsp_get_byte(unsigned int base) +{ + int val; + int i; + for (i = BUSY_LOOPS; i; i--) { + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + val = inb(SBP1(base, READ)); + return val; + } + } + return -ENODEV; +} + +static int my_sbdsp_reset(unsigned int base) +{ + int i; + + outb(1, SBP1(base, RESET)); + udelay(10); + outb(0, SBP1(base, RESET)); + udelay(30); + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + if (inb(SBP1(base, READ)) == 0xaa) + return 0; + else + break; + } + return -ENODEV; +} + +static int jazz16_irq_dma_setup(unsigned int base, uint sbirq, uint mpuirq, uint dma8, uint dma16) +{ + u8 dma_config, irq_config; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB irq %u, dma %u&%u, MPU irq %u.\n", + sbirq, dma8, dma16, mpuirq); + irq_config = jazz16_irq_config(sbirq) | jazz16_irq_config(mpuirq) << 4; + dma_config = jazz16_dma_config(dma8) | jazz16_dma_config(dma16) << 4; + if (!my_sbdsp_command(base, SB_JAZZ16_SET_DMAINTR)) + return 0; + + if (!my_sbdsp_command(base, dma_config)) + return 0; + jazz16_sb_dma1 = dma8; + jazz16_sb_dma2 = dma16; + + if (!my_sbdsp_command(base, irq_config)) + return 0; + jazz16_sb_irq = sbirq; + jazz16_mpu401_irq = mpuirq; + return 1; +} + +static int jazz16pnp_get_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + pnp_init_resource_table(res); + if (dev == jazz16_gameport_dev) { + res->port_resource[0].start = res->port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + res->port_resource[0].flags = IORESOURCE_IO; + } else if (dev == jazz16_sb_dev) { + if (!jazz16_sb_port) + return 0; + res->port_resource[0].start = jazz16_sb_port; + res->port_resource[0].end = jazz16_sb_port + 0x10 - 1; + res->port_resource[0].flags = IORESOURCE_IO; + + res->irq_resource[0].start = jazz16_sb_irq; + res->irq_resource[0].end = jazz16_sb_irq; + res->irq_resource[0].flags = IORESOURCE_IRQ; + if (jazz16_mpu401_irq) { + res->irq_resource[1].start = jazz16_mpu401_irq; + res->irq_resource[1].end = jazz16_mpu401_irq; + res->irq_resource[1].flags = IORESOURCE_IRQ; + } + + res->dma_resource[0].start = jazz16_sb_dma1; + res->dma_resource[0].end = jazz16_sb_dma1; + res->dma_resource[0].flags = IORESOURCE_DMA; + if (jazz16_sb_dma2 != jazz16_sb_dma1) { + res->dma_resource[1].start = jazz16_sb_dma2; + res->dma_resource[1].end = jazz16_sb_dma2; + res->dma_resource[1].flags = IORESOURCE_DMA; + } + res->port_resource[1].start = jazz16_mpu401_port; + res->port_resource[1].end = jazz16_mpu401_port + 2 - 1; + res->port_resource[1].flags = IORESOURCE_IO; + } else + return -EINVAL; + return 0; +} + +/* Disables the device if supported. */ +static int jazz16pnp_disable_resources(struct pnp_dev *dev) +{ + if (dev == jazz16_gameport_dev) { + return -ENOSYS; + } else if (dev == jazz16_sb_dev) { + if (jazz16_sb_port) + jazz16_irq_dma_setup(jazz16_sb_port, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + if (jazz16pnp_fm_res) { + release_resource(jazz16pnp_fm_res); + jazz16pnp_fm_res = NULL; + } + dev->active = false; + return 0; + } else { + return -EINVAL; + } +} + +#define PNP_RES_IS_SET(res,type,TYPE,i) \ + (((res)->type##_resource[i].flags & (IORESOURCE_##TYPE | IORESOURCE_UNSET)) \ + == IORESOURCE_##TYPE) + +static int jazz16pnp_set_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + unsigned int sbirq, mpuirq, dma1, dma2; + + if (dev == jazz16_gameport_dev) + return -ENOSYS; + + else if (dev == jazz16_sb_dev) { + if (PNP_RES_IS_SET(res, dma, DMA, 0)) + dma1 = res->dma_resource[0].start; + else + return -EINVAL; + if (PNP_RES_IS_SET(res, dma, DMA, 1)) + dma2 = res->dma_resource[1].start; + else + dma2 = dma1; /* "half duplex" operation. */ + if (PNP_RES_IS_SET(res, irq, IRQ, 0)) + sbirq = res->irq_resource[0].start; + else + return -EINVAL; + if (PNP_RES_IS_SET(res, irq, IRQ, 1)) + mpuirq = res->irq_resource[1].start; + else + mpuirq = 0; /* MPU-401 driver supports this. */ + if (! PNP_RES_IS_SET(res, port, IO, 0)) + return -EINVAL; + if (! PNP_RES_IS_SET(res, port, IO, 1)) + return -EINVAL; + + /* Block the OPL3 alias to avoid conflicts. */ + jazz16pnp_fm_res = request_region(0x388, 4, "SoundBlaster FM"); + /* Avoid setting the mpu port after setting the sb port + because doing so will glitch the mpu port from the + previous one to the new one. */ + jazz16_port_setup(res->port_resource[0].start, + res->port_resource[1].start); + jazz16_irq_dma_setup(jazz16_sb_port, sbirq, mpuirq, dma1, dma2); + dev->active = true; + return 0; + } else + return -EINVAL; +} +#undef PNP_RES_IS_SET + +static struct pnp_protocol jazz16pnp_protocol = { + .name = "Jazz16 PnP", + .get = jazz16pnp_get_resources, + .set = jazz16pnp_set_resources, + .disable = jazz16pnp_disable_resources, +}; + +static uint __init jazz16_check(unsigned int base) +{ + if (my_sbdsp_reset(base)) + return 0; + if (!(my_sbdsp_command(base, SB_DSP_GET_JAZZ_VERSION))) + return 0; + return SB_VERSION_IS_JAZZ16(my_sbdsp_get_byte(base)); +} + +static int __init jazz16pnp_init(void) +{ + int err = 0; + unsigned int base, mpubase; + + if (jazz16pnp_disable == 1) { + printk(PFX_INFO "Jazz16 Plug & Play support disabled.\n"); + return 0; + } + if (!request_region(SB_JAZZ16_CONFIG_PORT, 1, "Jazz16 PnP probe")) + return -EBUSY; + + for (mpubase = 0x300; mpubase <= 0x330; mpubase += 0x10) + if (request_region(mpubase, 2, "Jazz16 PnP probe")) + break; + if (mpubase > 0x330) + goto out_no_mpubase; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Scanning for Jazz16 cards...\n"); + + for (base = 0x260; base >= 0x210; base -= 0x10) { + if (!request_region(base, 16, "Jazz16 PnP probe")) + continue; + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Probing at %#x (with MPU at %#x).\n", + base, mpubase); + jazz16_port_setup(base, mpubase); + if (jazz16_check(base)) + jazz16pnp_count++; + jazz16_irq_dma_setup(base, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + release_region(base, 16); + if (jazz16pnp_count) + break; + } + if (base < 0x210) + goto out_no_sbbase; + if (!jazz16pnp_count) { + err = -ENODEV; + goto out_no_devices; + } + if (pnp_register_protocol(&jazz16pnp_protocol) < 0) + err = -ENOMEM; + else { + if (jazz16pnp_add_gameport()) + printk(PFX_WARN "Failed to add game port.\n"); + if (jazz16pnp_add_sb()) + printk(PFX_WARN "Failed to add SB port.\n"); + err = 0; + } +out_no_devices: + printk(PFX_INFO "%u Jazz16 card%s detected.\n", jazz16pnp_count, + jazz16pnp_count == 1 ? "" : "s"); +out_no_sbbase: + release_region(mpubase, 2); +out_no_mpubase: + release_region(SB_JAZZ16_CONFIG_PORT, 1); + return err; +} +module_init(jazz16pnp_init); + +static void __exit jazz16pnp_exit(void) +{ + pnp_unregister_protocol(&jazz16pnp_protocol); +} +module_exit(jazz16pnp_exit); -- Rask Ingemann Lambertsen

3 8

[alsa-devel] 2.6.21-rc7: known regressions with patches
by Adrian Bunk 26 Apr '07

26 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20 with patches available. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available Subject : qla2xxx driver no longer works References : http://lkml.org/lkml/2007/4/16/49 http://lkml.org/lkml/2007/4/20/224 Submitter : David Miller <davem(a)davemloft.net> Robert Peterson <rpeterso(a)redhat.com> Handled-By : David Miller <davem(a)davemloft.net> Patch : http://lkml.org/lkml/2007/4/16/234 Status : patch was suggested

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[alsa-devel] playback open error:
by Ashlesha Shintre 25 Apr '07

25 Apr '07

Hi, I m trying to run the simple playback program that I have attached with this email -- and I get the following error: ALSA lib pcm_dmix.c:862:(snd_pcm_dmix_open) unable to open slave > Playback open error: Device or resource busy I am running this program as a function called wavplay (attached) -- however, if i run the same program, as part of the main function, then there is no problem -- What i mean is that if i basically just paste the code for the wavplay function in main and provide the input values (the pcm data to be played) -- then it can very well open the default device and play the pcm data that is being passed to it! Any pointers? Thanks, Ashlesha.

2 2

[alsa-devel] daily snapshot & version ID.
by Tobin Davis 25 Apr '07

25 Apr '07

Would it be possible to add a version ID update to the daily snapshots that you generate? It would make tracking a little easier, and we could get more testing on recent snapshots. That or spin another RC soon. -- Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] crash/reboot with rawmidi on ice1712 dual opteron
by Florian 25 Apr '07

25 Apr '07

Hi, on one particular machine, an IBM workstation (Intellistation A Pro with dual opteron 250, 4GB RAM), opening a rawmidi input port on an M-Audio 2496 card will cause an immediate reboot of the machine. I've tried with different SMP kernels and different ALSA versions (including 1.0.14rc3), but always the same. Syslog does not show anything. Using a USB MIDI interface works fine. Also, MIDI input on a similar workstation (Intellistation Z Pro, dual xeon, 1GB RAM) with M-Audio 2496 works without problems. Any ideas? Thanks, Florian -- Florian Bomers bome.com ------------------------------------------------------- Music Software, Development Tools: http://www.bome.com Java Sound extensions, plugins: http://www.tritonus.org The Java Sound Resources: http://www.jsresources.org ------------------------------------------------------- Please quote this email in your reply. Thanks!

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[alsa-devel] Digigram ISA cards
by Ash Willis 25 Apr '07

25 Apr '07

Hi, I'm thinking about trying to write a driver for some old Digigram cards soon, namely the PCX11 and PCX9. I might be wrong but it seems these cards natively support MPEG layers I and II and only actually support PCM by re-encoding in-driver. Does alsa-lib support MPEG conversions? Are there examples of drivers that already do this type of thing? Please forgive my ignorance. I grepped the sources and didn't find too much info. Ash =

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[alsa-devel] [1/3] 2.6.21-rc7: known regressions (v2)
by Adrian Bunk 25 Apr '07

25 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : gammu no longer works References : http://lkml.org/lkml/2007/4/20/84 Submitter : Wolfgang Erig <Wolfgang.Erig(a)gmx.de> Status : unknown Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Wrong "From" email address selected earlier. Oops.

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

2 1

[alsa-devel] ICE 1724 MIDI support still broken?
by Daniel James 24 Apr '07

24 Apr '07

Hello ALSA team, We have a user with an ESI Juli@ whose Linux box hangs when any MIDI application starts up. I read that MIDI support is broken generally on ICE 1724 cards; is this still the case in recent driver releases? If so, is there any interest in fixing this long-standing issue? The user has indicated that his box would be available by ssh for any tests required. Cheers! Daniel

2 3

[alsa-devel] Standard config and plugin paths
by Takashi Iwai 24 Apr '07

24 Apr '07

Hi, while working on openSUSE 10.3 ALSA packaging, I had a discussion regarding the config and plugin paths. Basically, the current configs (typically in /usr/share/alsa or /etc/alsa) and binary plugins (/usr/lib/alsa-lib) are fairly strictly bound to the specific library, libasound.so.2. That is, if we publish a newer, incompatible version, libasound.so.3, these files will be very unlikely unusable, too. The suggestion was that we should keep these config and plugin files in the version-dependent paths so that no conflictions would occur at update. For example, instead of /usr/share/alsa, /usr/share/alsa2 or whatever. The fix would be easy -- just add some options to configure script of alsa-lib and alsa-plugins. So, the question is what path names are most suitable as the standard. For example, I find the following OK: - $DATADIR/alsa-1.0/ - $LIBDIR/alsa-lib-1.0/ Or, the deeper paths: - $DATADIR/alsa/1.0/ - $LIBDIR/alsa-lib/1.0/ For keeping the backward compatibility, we may implement fallback to read the old default path in addition, if the new path doesn't exist. In that case, the former suggestion would be cleaner. Any comments or suggestions? Takashi

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[alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 24 Apr '07

24 Apr '07

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] What does start_threshold mean, and what value should it contain?
by j t 24 Apr '07

24 Apr '07

Hi. I'm trying to debug a problem with xruns which happen when playing a wav file with vlc, but I can't reproduce the problems when using "aplay -M" to play the same wav file. (logged as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3031). I've checked the contents of /proc/asound/card0/pcm0p/sub0/ in both cases - they appear to be very similar except for the contents of sw_params: when I use vlc, start_threshold says 1, but when I use "aplay -M", start_threshold says 16384. So, what does start_threshold mean, is there a particular value that start_threshold "should" contain, and could this difference be the cause of the xruns? (The aplay man page doesn't seem to have any flags for controlling the value of this parameter, so it doesn't appear to be "user-configurable"). (Apologies if this is the wrong mailing list - I though that -devel would be better than -user, but I could be wrong). Thanks, Jaime

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Re: [alsa-devel] Fwd: No sound with ATI SB450
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 22:45:00 +1000, Ben DiDonc wrote: > > Tobin, Takashi, > Thank you very much for your help. > Takashi's last suggestion did it and I finally got a working ATI SB450 > sound card with Ubtuntu Feisty (kernel 2.6.20 and > alsa-driver-hg20070421). I did a (last) alsa-info.sh in case it helps > others, see http://pastebin.ca/454115 > Here's what it took: > modprobe snd_hda_intel probe_mask=8 model=auto > Without probe_mask, the ALC861 is not properly detected. Without > model=auto, playback doesnt work. > Let me know If I can be of any help or if you want me to try something else. As I requested, get /proc/asound/card0/codec#* files for both working and non-working states. Then we can compare what is not set up properly. probe_mask is a bit hard to fix right now. A least, we need to know which modem codec is harmful. Please check the codec#* file again with probe_mask=1. Takashi > Thanks again. > Ben > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 23 Apr 2007 21:58:17 +1000, > > Ben DiDonc wrote: > > > > > > Tobin, Takashi, > > > upon Takashi's advice, I tried loading hda-intel > > > (alsa-driver-hg20070421) using the probe_mask argument under 2.6.20. > > > With probe_mask=1, the codec isnt detected and only the modem shows up > > > (just like before). However, with probe_mask=8, the codec IS correctly > > > detected (alsa-info.sh at http://pastebin.ca/454059 ). syslog shows: > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 19 > > > ALSA .../alsa-driver-hg20070421/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:696: > > > codec_mask = 0x9 > > > ALSA .../alsa/alsa-driver-hg20070421/pci/hda/hda_codec.c:1819: > > > hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) > > > > > > However, there is still no playback (no muted control, all volumes at > > > 100% as shown by amixer in alsa-info.sh). Here again, capture works, I > > > successfully captured from the mic with arecord. > > > > Now you can try what I suggested, model=auto. And compare with > > codec#* files. If you have /proc/asound/card0/codec#* files on the > > working system, it'd be better, in addition... > > > > > > Takashi > > > > > > > > Can I draw your attention on something ? I m not sure about what i m > > > going to say (specially the part on NIDs) but i ll try. I know Tobin > > > already explained that this was normal, but if you have a look at the > > > post where I loaded edgy (2 posts ago), where the playback > > > works(2.6.17 with 1.0.12), you ll see that when I aplay something, > > > syslog shows only a single call to hda_codec_setup_stream with > > > NID=0x3. In all cases where playback doesnt work (1.0.14rcX under > > > 2.6.20), running aplay makes 4 calls to hda_codec_setup_stream with > > > NID=3, 6 ,5 and 4 in turn. Why does it try 6 5 and 4 ? Is it because 3 > > > fails ? We know that 3 should be used as its the one that works (under > > > 2.6.17). Now arent NIDs node IDs of codecs connected to the HDA > > > controller ? Is it possible that 6, 5 and 4 are mistakenly detected > > > and cannot be used for playback ? > > > Let me know if I m way off track. > > > > > Ben > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Mon, 23 Apr 2007 11:05:19 +0000, > > > > Ben DiDonc wrote: > > > > > > > > > > Takashi, > > > > > I did try passing model=auto to all 14 release candidates, > > > > > > > > Did you already try the HG version (or daily snapshot), right? > > > > > > > > > but without > > > > > success. the codec is never detected (see my previous posts for > > > > > alsa-info.sh outputs with both 2.6.20 and 2.6.17). > > > > > My problem is I cant get my sound card to work with kernel 2.6.20, > > > > > when it used to work with 2.6.17. > > > > > > > > The codec mask probed is 0x09, so you can pass either probe_mask=8 or > > > > probe_mask=1 to skip the modem detection (if it matters). > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > Ben > > > > > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > At Mon, 23 Apr 2007 10:50:18 +0000, > > > > > > Ben DiDonc wrote: > > > > > > > > > > > > > > Tobin, > > > > > > > Unfortunately, ACPI was on at all times. My laptop hardly ever boots > > > > > > > without ACPI (randomly, the SATA driver fails before it can even mount > > > > > > > the root partition). > > > > > > > I ran some more tests today: > > > > > > > I loaded Ubuntu Edgy from the live CD where the sound chip used to > > > > > > > work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works > > > > > > > straight away. From there , I ran the same series of test: download > > > > > > > and build (--with-debug=detect) the next version of alsa-driver, run a > > > > > > > capture from mic and try playing it back, check syslog and run > > > > > > > alsa-info.sh. I did this for successive versions of alsa-driver > > > > > > > (starting with 1.0.12) until it fails. > > > > > > > Here are the results: > > > > > > > > > > > > > > Alsa driver 1.0.12rc1(shipped with edgy): > > > > > > > capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 > > > > > > > > > > > > > > Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) > > > > > > > syslog lines after modprobe: > > > > > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: > > > > > > > codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: > > > > > > > line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 > > > > > > > (0x0/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, > > > > > > > dig_out=0x0, din_in=0x0 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, > > > > > > > fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 > > > > > > > > > > > > > > capture: working > > > > > > > //capture starting > > > > > > > hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, > > > > > > > channel=0, format=0x11 > > > > > > > //capture stopping > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > playback: working > > > > > > > //starting > > > > > > > ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, > > > > > > > fragsize=0x1000, format=0x11 > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, > > > > > > > channel=0, format=0x11 > > > > > > > //stopping > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) > > > > > > > modprobe snd-hda-intel: > > > > > > > hda_intel.c:680: codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, > > > > > > > cd=0x11, aux=0x0 > > > > > > > capture works fine, syslog output similar to 1.0.12 plus an extra > > > > > > > "hda-intel: Invalid position buffer, using LPIB read method instead" > > > > > > > in the middle of the capture. Playback works fine too, with the same > > > > > > > syslog entries as 1.0.12 > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) > > > > > > > modprobe: > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works (output similar to 1.0.12) > > > > > > > Playback does NOT work this time: > > > > > > > syslog shows a call to azx_pcm_prepare: and 4 calls to > > > > > > > hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out > > > > > > > my previous post) > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) > > > > > > > modprobe: the codec is correctly detected > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works. Playback does NOT work, same syslog entries as > > > > > > > 1.0.14rc1 with multiple calls to hda_codec_setup_stream > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) > > > > > > > modprobe: the codec isnt detected. > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > capture and playback dont work. > > > > > > > > > > > > > > Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy > > > > > > > (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but > > > > > > > not playback, 14rc3 doesnt detect the codec. > > > > > > > What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel > > > > > > > between releases 13 and 14 ... > > > > > > > > > > > > Try to pass model=auto to snd-hda-intel module, and check whether it > > > > > > works (make sure that you have unmuted/adjusted mixer elements before > > > > > > that). Then get /proc/asound/card0/codec#* files, and compare with > > > > > > the files without model option. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > > > >

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[alsa-devel] Compilation of latest alsa-hg
by dustin＠seznam.cz 24 Apr '07

24 Apr '07

Hello, please could you point me at a functioning description of latest complete hg tree compilation? The guides e.g. http://www.suseforums.net/index.php?showtopic=30156&pid=170707&st=0&#entry1… end up with /home/pavel/tmp/alsa-hg/alsa/alsa-driver/soc/codecs/../../alsa-kernel/soc/codecs/wm8753.c:1707: error: ‘I2C_DRIVERID_WM8753’ undeclared here (not in a function) Thanks a lot. Pavel.

2 1

Re: [alsa-devel] Functioning support for Prodigy 192
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 14:00:22 +0200 (CEST), dustin(a)seznam.cz wrote: > > Hello, > > Back here again :) > > While comparing media reading precision of various CD players I hit the following issues: > > 1. De-emphasis filter in AK4114 is turned on, distorting incoming data. A patch to disable: > > diff -r aed6a15c9398 pci/ice1712/prodigy192.c > --- a/pci/ice1712/prodigy192.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/prodigy192.c Mon Apr 23 12:47:01 2007 +0200 > @@ -714,7 +714,7 @@ static int prodigy192_ak4114_init(struct > { > static const unsigned char ak4114_init_vals[] = { > AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1, > - AK4114_DIF_I24I2S, /* ice1724 expects I2S and provides clock */ > + AK4114_DIF_I24I2S | AK4114_DEM0 , /* ice1724 expects I2S and provides clock, DEM0 disables deemphasis filter */ > AK4114_TX1E, > AK4114_EFH_1024 | AK4114_DIT, /* default input RX0 */ > 0, Looks OK. > 2. All data recorded when master clock was provided by AK4114 (Multi Track Internal Clock= IEC958 Input) had double the sampling rate. For each input sample there were two identical output samples in the file. > > Here is where the FPGA on MIODIO gets introduced. Both MCKO1 and MCKO2 of ak4114 are fed to FPGA, which outputs master clock to SPMCLKIN of ice1724. Unfortunately my oscilloscope has just died, so experimentally I found out that only a combination of OCKS0=1, OCKS1 = 1 (128fs, 64fs output) and ice1724- VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct sampling rate. In order for this to work the FPGA must double the MCK01 rate (I cannot check the frequency now). > > It may actually make sense as it is probably the only way to support input sampling frequencies all the way up to 192kHz (128fs is max ak4114 can output at 192kHz) without having to switch the VT1724_MT_I2S_MCLK_128X bit in ice1724. In consumer mode the chips have no information about the sampling frequency (ak4114 does no detection) and the driver would not know when to set the bit. > > The problem is that due to switching sequence of Multi Track Internal Clock enum the VT1724_MT_I2S_MCLK_128X stays set (from the previous 192kHz option). I think it should not be set for ice1724 slave mode at all, here is the patch which fixes the sampling rate. Looks fine, too. Could you post a patch with a short patch summary and sign-off to apply to the upstream tree? thanks, Takashi > > Thanks, > > Pavel. > > diff -r aed6a15c9398 pci/ice1712/ice1724.c > --- a/pci/ice1712/ice1724.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/ice1724.c Mon Apr 23 13:57:19 2007 +0200 > @@ -1666,7 +1666,11 @@ static int snd_vt1724_pro_internal_clock > spin_lock_irq(&ice->reg_lock); > oval = inb(ICEMT1724(ice, RATE)); > if (ucontrol->value.enumerated.item[0] == spdif) { > + unsigned char i2s_oval; > outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); > + /* setting 256fs */ > + i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); > + outb( i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); > } else { > rate = rates[ucontrol->value.integer.value[0] % 15]; > if (rate <= get_max_rate(ice)) { > >

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[alsa-devel] Buffer Overrun Messages
by John Dey 24 Apr '07

24 Apr '07

Hi, I have been driving fluidsynth with with aplaymidi and get the following: fluidsynth: ALSAsequencer buffer overrun, lost event. I have played around with buffer size and buffer number and also run with and without jackd. The sound out doesn't seem to be affect. Can anyone give me some guidance on how to eliminate the error. Could the error be caused by fluidsynth? I have also run the program on PS2/Yellowdog linux and CCRMA fc5 (x86) both with similar results. Thanks. John

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[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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Re: [alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, Unfortunately, ACPI was on at all times. My laptop hardly ever boots without ACPI (randomly, the SATA driver fails before it can even mount the root partition). I ran some more tests today: I loaded Ubuntu Edgy from the live CD where the sound chip used to work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works straight away. From there , I ran the same series of test: download and build (--with-debug=detect) the next version of alsa-driver, run a capture from mic and try playing it back, check syslog and run alsa-info.sh. I did this for successive versions of alsa-driver (starting with 1.0.12) until it fails. Here are the results: Alsa driver 1.0.12rc1(shipped with edgy): capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) syslog lines after modprobe: ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, dig_out=0x0, din_in=0x0 ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture: working //capture starting hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 //capture stopping hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 playback: working //starting ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 //stopping ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) modprobe snd-hda-intel: hda_intel.c:680: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture works fine, syslog output similar to 1.0.12 plus an extra "hda-intel: Invalid position buffer, using LPIB read method instead" in the middle of the capture. Playback works fine too, with the same syslog entries as 1.0.12 alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) modprobe: ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works (output similar to 1.0.12) Playback does NOT work this time: syslog shows a call to azx_pcm_prepare: and 4 calls to hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out my previous post) alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) modprobe: the codec is correctly detected ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works. Playback does NOT work, same syslog entries as 1.0.14rc1 with multiple calls to hda_codec_setup_stream alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) modprobe: the codec isnt detected. ALSA hda_intel.c:696: codec_mask = 0x9 capture and playback dont work. Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but not playback, 14rc3 doesnt detect the codec. What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel between releases 13 and 14 ...

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[alsa-devel] ad1986a snd_hda_intel patch
by Dmitry Ursegov 23 Apr '07

23 Apr '07

Hello! I have an asus a8jn laptop and a integrated audio with Analog Devices AD1986A chip. $ lspci -vnn 0:1b.0 Audio device [0403]: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: ASUSTeK Computer Inc. Unknown device [1043:1447] Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at febfc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> The audio seems to work properly on the right channel, but not on the left. All I get on the left channel is a loud screeching noise followed by a pop when the audio turns off. The microphon jack don't work also. I make a patch from the solution described in this http://mailman.alsa-project.org/pipermail/alsa-devel/2007-March/000093.html message: diff -Nur alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c --- alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c 2007-03-06 15:26:32.000000000 +0300 +++ alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c 2007-04-20 23:32:19.000000000 +0400 @@ -873,7 +873,7 @@ board_config = snd_hda_check_board_config(codec, AD1986A_MODELS, ad1986a_models, ad1986a_cfg_tbl); - switch (board_config) { +/* switch (board_config) { case AD1986A_3STACK: spec->num_mixers = 2; spec->mixers[1] = ad1986a_3st_mixers; @@ -912,6 +912,18 @@ spec->multiout.dig_out_nid = 0; break; } +*/ + + spec->num_mixers = 2; + spec->mixers[1] = ad1986a_3st_mixers; + spec->num_init_verbs = 3; + spec->init_verbs[1] = ad1986a_3st_init_verbs; + spec->init_verbs[2] = ad1986a_ch2_init; + spec->channel_mode = ad1986a_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); + spec->need_dac_fix = 1; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; return 0; } I don't know, why snd_hda_check_board_config() return wrong value, but that workaround works. Thanks! Dmitry Ursegov

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[alsa-devel] Strange code in
by Pete Zaitcev 23 Apr '07

23 Apr '07

Dear Takashi: Hans de Goede (on cc:) hits a problem when snd-usb-audio breaks with: "USB device is in the shutdown state, cannot create a card instance". This is how the place looks: chip = NULL; for (i = 0; i < SNDRV_CARDS; i++) { if (usb_chip[i] && usb_chip[i]->dev == dev) { if (usb_chip[i]->shutdown) { snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n"); goto __error; } chip = usb_chip[i]; break; } } The code looks broken to me, because dev at this time is a new device structure, freshly allocated. You cannot compare it with an old instance, and I don't know why you would want to. Unfortunately (for fortunately), I have zero clue about the way ALSA works, specifically rules for snd_device_ops.dev_free, so I am asking you to look at this problem. Yours, -- Pete

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[alsa-devel] [PATCH] sound: fix incorrect use of platform_device_register()
by Akinobu Mita 23 Apr '07

23 Apr '07

The platform_device allocated by platform_device_alloc() should be added to the device hierarchy by platform_device_add() instead of platform_device_register(). Otherwise it will hit WARN_ON() in platform_device_register(). by illegal refcount. This patch fixes such incorrect usages in portman2x4 and mts64 drivers. Also it removes unnecessary trailing whitespaces. Cc: Jaroslav Kysela <perex(a)suse.cz> Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Index: 2.6-mm/sound/drivers/portman2x4.c =================================================================== --- 2.6-mm.orig/sound/drivers/portman2x4.c +++ 2.6-mm/sound/drivers/portman2x4.c @@ -676,13 +676,13 @@ static void __devinit snd_portman_attach struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; } Index: 2.6-mm/sound/drivers/mts64.c =================================================================== --- 2.6-mm.orig/sound/drivers/mts64.c +++ 2.6-mm/sound/drivers/mts64.c @@ -892,13 +892,13 @@ static void __devinit snd_mts64_attach(s struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; }

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