June 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>

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[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

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Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Takashi Iwai 09 Jul '07

09 Jul '07

At Mon, 25 Jun 2007 15:16:41 -0500, Matt Mullins wrote: > > On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > Yes, single_cmd mode is used always. > > That's bad. > > I believe we have established that. What can I do about it? *Can* I > do anything about it? What functions should I start debugging? Has > this happened with other codecs before? What documentation will I > need or want other than the Intel HDA and STAC9205 specifications? Similar problems have been seen in some laptops, too. I vaguely remember that they need some ACPI-related boot options to fix it somehow. I think it's likely irrelevant with the codec chip, but rather a problem of the controller side. > I'm definitely willing to help work on this, as not many people have > this hardware. I'm just new to the sound system, so I'm still trying > to figure out what-calls-what. Takashi

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[alsa-devel] [HDA Intel/STAC92xx] mic not working (retasking?)...
by Daniel J Blueman 09 Jul '07

09 Jul '07

With both 2.6.20 and 2.6.22 kernels on Ubuntu 7.04 on my Sony Vaio SZ240, I'm unable to get my mic connector working at any cost. Well, almost...reading from /dev/dsp, I do see a minute change from 127 for 126 or 128 if I tap my external mic. This would imply the jack-retasking feature of this darn STAC chipset, right? How can I change the jack retasking with the Intel HDA drivers or get further insight into this problem? Thanks in advance, Daniel --- [1] $ lspci -s0:1b.0 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ amixer info Card default 'Intel'/'HDA Intel at 0xde400000 irq 22' Mixer name : 'SigmaTel CXD9872RD/K' Components : 'HDA:83847661 HDA:14f12bfa' Controls : 6 Simple ctrls : 5 $ amixer controls numid=2,iface=MIXER,name='Master Playback Switch' numid=1,iface=MIXER,name='Master Playback Volume' numid=6,iface=MIXER,name='PCM Playback Volume' numid=5,iface=MIXER,name='Capture Source' numid=4,iface=MIXER,name='Capture Switch' numid=3,iface=MIXER,name='Capture Volume' $ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'PCM',0 Simple mixer control 'Line',0 Simple mixer control 'Mic',0 Simple mixer control 'Capture',0 $ amixer contents numid=2,iface=MIXER,name='Master Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=1,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=127,step=0 : values=66,66 numid=6,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=230,230 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=5,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Line' ; Item #1 'Mic' : values=1 numid=4,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=3,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=15,step=0 : values=15,15 | dBscale-min=0.00dB,step=1.50dB,mute=0 -- Daniel J Blueman

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[alsa-devel] PCMCIA Audigy 2 broken in 2.6.21?
by Chuck Ebbert 04 Jul '07

04 Jul '07

Multiple users report the PCMCIA Audigy 2 adapter hangs the machine in 2.6.21 while it work OK in 2.6.20: https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=242208

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[alsa-devel] SALSA-Lib: snd_pcm_hw_params fails with MMAP
by J. Scott Merritt 03 Jul '07

03 Jul '07

Dear List, Using SALSA-Lib 0.0.5 ... After calling snd_pcm_hw_params_set_access with an access mode of SND_PCM_ACCESS_MMAP_INTERLEAVED, the subsequent call to snd_pcm_hw_params returns a snd_strerror error code of "Invalid argument". This same program operates properly in the target environment (ARM PXA270) when compiled and linked with the standard ALSA LIB 1.0.13. Thanks, Scott.

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[alsa-devel] roland synth sh-201
by carlo beccaria 03 Jul '07

03 Jul '07

hi all, with this change in usbquirks.h , i can use my synth roland sh-201 { /* Roland SH-201 */ USB_DEVICE(0x0582, 0x00ad), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Roland", .product_name = "SH-201", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x01, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x81, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } }, my kernel is 2.6.21.5-rt17, (plus preempt patch) and i use debian testing the hardware seem to be very similar to Roland SC-D70 (windows .inf file are very similar) this is the device id Bus 001 Device 003: ID 0582:00ad Roland Corp. in attach you can see the lsusb (verbose) output becks@vaix:~$ amidi -l Dir Device Name IO hw:1,0,0 SH-201 MIDI 1 IO hw:1,0,1 SH-201 MIDI 2 IO hw:1,0,2 SH-201 MIDI 3 muse works fine with jack and midi output becks@vaix:~$ aseqdump -p 20:0 Waiting for data. Press Ctrl+C to end. Source_ Event_________________ Ch _Data__ 20:0 Note on 1 52 17 20:0 Note off 1 52 55 also midi-in works becks@vaix:~$ cat /proc/asound/cards 0 [A5451 ]: ALI5451 - ALI 5451 ALI 5451 at 0x8400, irq 9 1 [SH201 ]: USB-Audio - SH-201 Roland SH-201 at usb-0000:00:0c.0-1, full speed when i use my synth like a sound card, it don't work, i don't know exactly why (if is a patch problem, or rate/format) becks@vaix:~$ aplay -D hw:1,0 /usr/share/sounds/KDE_Window_UnSticky.wav Playing WAVE '/usr/share/sounds/KDE_Window_UnSticky.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono aplay: set_params:904: Sample format non available let me know if you need more info thanks -- carlo beccaria

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[alsa-devel] [HDA-Intel/ALC268] no sound
by Steven Zakulec 02 Jul '07

02 Jul '07

(Continued from this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001459.html) I've included the result of alsactl -f somefile store. I thought it got sent to the list earlier, but apparently not. Below this I also added the alsa things from dmesg. state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Front Playback Volume' value.0 57 value.1 57 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Headphone Playback Volume' value.0 63 value.1 63 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' value.0 1 value.1 1 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 false value.1 false } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' index 1 value.0 1 value.1 1 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.9 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' value Mic } control.10 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' index 1 value Mic } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Off-hook Switch' value false } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Caller ID Switch' value false } control.13 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } } hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 19.236000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 19.244000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) comment.count 2 :

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[alsa-devel] [HDA Intel/STAC92xx] Alsa 1.0.14: error in stac92xx_auto_fill_dac_nid
by Jan de Mooij 02 Jul '07

02 Jul '07

I have a strange problem with alsa on Ubuntu Gutsy. If I compile and install the Alsa 1.0.14 driver, my sound card is not detected, and sound doesn't work. I also get the following message in dmesg: [ 33.360190] stac92xx_auto_fill_dac_nids: No available DAC for pin 0x15 So the problem seems to be in the stac92xx_auto_fill_dac_nids function. When I copy that function from alsa 1.0.13 into the 1.0.14 source, everything works okay... It looks like the function is rewritten for version 1.0.14, but for me it does not work anymore. I have to do the above steps with each new kernel update in Ubuntu Gutsy... Therefore I would like to fix it, but I don't know where to start. Thanks in advance. Jan

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[alsa-devel] Acer Aspire 9810 hda-intel RealTek ALC883 no sound thru speakers only via SPDIF/Headphones.
by Adrian Grassi 02 Jul '07

02 Jul '07

Hi guys, I have a Acer Aspire 9810 with hda-intel realtek ALC883 chipset and no matter what version of alsa and ditribution I try I only get sound thru the shared SPDIF/Headphone Jack. This laptop has a built in 2.1 power speaker system which works fine under windows but not in any linux distro's I have tried: Ubuntu 7.04 - Kubuntu 7.04 - Suse 10.2 - Ubuntu Gutsy 7.10 Beta - Mandriva Spring 2007.1 I think this is a configuration issue however I can't find any info anywhere on how to fix it, I did find a few posts where several people are having the same issue. Can someone please shed some light on how to fix this issue, if any additional information is needed please let me know, I have been trying to solve this for over 5 days now and is driving me bananas. Here is the aadebug from alsa-wiki: Debug INFO ALSA Audio Debug v0.1.0 - Sat Jun 30 02:01:40 EDT 2007 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux XTTG-Mob1 2.6.22-7-generic #1 SMP Mon Jun 25 17:33:14 GMT 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 256024 1 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 3 snd_hda_intel,saa7134_alsa,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 2 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 12 snd_hda_intel,saa7134_alsa,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 10888 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xd2300000 irq 23 1 [SAA7134 ]: SAA7134 - SAA7134 saa7133[0] at 0xd2005000 irq 16 2: : timer 3: : sequencer 4: [ 1- 0]: digital audio capture 5: [ 1] : control 6: [ 0- 2]: digital audio capture 7: [ 0- 0]: digital audio playback 8: [ 0- 0]: digital audio capture 9: [ 0] : control cat: /proc/asound/hwdep: No such file or directory 01-00: SAA7134 PCM : SAA7134 PCM : capture 1 00-02: ALC883 Analog : ALC883 Analog : capture 2 00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 controlC1 pcmC0D0c pcmC0D0p pcmC0D2c pcmC1D0c seq timer CPU ------------------------------------------------------- model name : Intel(R) Core(TM)2 CPU T7200 @ 2.00GHz cpu MHz : 2000.000 model name : Intel(R) Core(TM)2 CPU T7200 @ 2.00GHz cpu MHz : 2000.000 RAM ------------------------------------------------------- MemTotal: 2074528 kB SwapTotal: 1951888 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corporation Mobile 945GM/PM/GMS/940GML and 945GT Express Memory Controller Hub (rev 03) 09:04.0 Multimedia controller: Philips Semiconductors SAA7133/SAA7135 Video Broadcast Decoder (rev d1) 09:06.2 Mass storage controller: Texas Instruments 5-in-1 Multimedia Card Reader (SD/MMC/MS/MS PRO/xD)

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[alsa-devel] High-pitched noise on output from cmi9739 (snd_intel8x0)
by j t 02 Jul '07

02 Jul '07

Hi all. I have a cmi9739 (sis7012) built into an asus k7s8x motherboard which produces a quiet but noticeable noise on top of the regular output (the noise only occurs during playback). I logged a bug to alsa's bugtrack: (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2288) about a year ago, but unfortunately I still have the problem with alsa release 1.0.14 To try to find out what's going on, I tried using seven (7) different drivers for this soundcard (both Linux and Windows XP) to find out when the problem occurs. Here's the results: a) Alsa (snd_intel8x0) version 1.0.14 under Ubuntu 7.10 alpha 2. Problem EXISTS. b) OSS 4Front Opensound v4.1 (v4.1test0-070624). Problem EXISTS c) CMedia Linux OSS driver cmaudio-070 (from http://www.cmedia.com.tw). Problem EXISTS. d) Windows XP "built-in" driver for SiS 7012 version 5.10.0.6120. Problem EXISTS. e) Windows XP drivers from Asus website for SiS 7012 version 5.12.1.38. Problem DOES NOT EXIST. f) Windows XP drivers from CMedia website (http://www.cmedia.com.tw) version 5.12.1.39. Problem DOES NOT EXIST. g) Windows XP drivers from CMedia forum (http://www.cmedia.com.tw/forums) version 5.12.1.51. Problem DOES NOT EXIST. Given that the problem does not happen with three out of the seven drivers, I assume it's a driver problem. Is there anything that I can do debug this further? Thank you.

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[alsa-devel] NVidia AC97 / CK804: arecord hangs in system call poll()
by Rainer Menzner 02 Jul '07

02 Jul '07

Hi, my soundcard is an onboard AC97 CK804 NVidia. I use snd-intel8x0 module. Everything works fine except recording sound. When calling arecord -f cd buffer.wav the program infinitely hangs, and an strace shows that it is a poll() call that does not return. After interrupting, a wave file of 44 bytes which is the header only is left. My setup is Kernel 2.6.21.3, all ALSA-software is updated to the latest version, that is 1.0.14(a). In a previous setup with an older ALSA version and kernel 2.6.16.11 recording did work with the same hardware. Now that I updated all ALSA software, even with that older kernel version the same recording problem came up. From that I concluded that it is somehow related to the ALSA-lib version. I have tried to downgrade ALSA-lib down to 1.0.0 to no avail. I have also made tests appropriate if this would be an interrupt problem (using pci=noroute et al) which also did not solve the problem. I have scrolled though the archives of alsa-user and there have been a few posts regarding the same problem (but with other chip sets). However, in those threads no definitive solution of the problem has been discussed. So I tried it again in alsa-user but did not get any feedback ... Now, I finally try this developer list. Your comments and help are much appreciated. Let me know if you need further info. I'm not a Linux system programmer or specialist, but I would be willing to do some debugging if somebody explains what to check. Thanks for any support, -Rainer

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[alsa-devel] hda-Intel Realtek ALC883 chipset support
by Adriano Grassi 30 Jun '07

30 Jun '07

Hi guys I have sound on different linux distro's however only the spdif/headphone share output, no sound from the internal speakers I have been trying to get this work for the last 5 days no success. So i was wondering if the support for this chipset is in place. I even tried the lastest candidate release alsa files 14rc4 with no luck, this is a high definition audio card with multiple outputs analog and digital and i have tried everything i can think of with no success, I tried multiple distro's with the same result, I don't this it is a driver issue is probably more like a configuration issue. Can anyone help me figure this one out. Thanks. 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xd2300000 irq 98 2: : timer 3: [ 0- 2]: digital audio playback 4: [ 0- 1]: digital audio capture 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0] : control 8: : sequencer cat: /proc/asound/Intel: Is a directory 0 snd_hda_intel cat: /proc/asound/oss: Is a directory 00-02: ALC883 Digital : ALC883 Digital : playback 1 00-01: ALC883 Analog : ALC883 Analog : capture 2 00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 2 cat: /proc/asound/seq: Is a directory G0: system timer : 4000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-0-3: PCM capture 0-0-3 : SLAVE P0-1-1: PCM capture 0-1-1 : SLAVE P0-1-3: PCM capture 0-1-3 : SLAVE P0-2-0: PCM playback 0-2-0 : SLAVE Advanced Linux Sound Architecture Driver Version 1.0.12rc1 (Thu Jun 22 13:55:50 2006 UTC).

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[alsa-devel] Digigram VX source code
by Colin Stewart Campbell 30 Jun '07

30 Jun '07

About 5 years ago Digigram released their VX soundcard source for OSX through the alsa project. Is there somewhere I can still get a hold of it? I have in mind a legacy card coreaudio driver for Intel macs. Colin

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[alsa-devel] helping debugging
by Thierry Vignaud 30 Jun '07

30 Jun '07

Hi Here's a suggestion from Dave Airlie: http://airlied.livejournal.com/46087.html "If you are going to have lots of unsigned int sized free running pointers called things like hw_ptr or hw.ptr or anything, for debugging it would be really cool if you started those counters at INT_MAX-100000 or something instead of at zero, 24 hours waiting for a bug to trip in the pcm rate conversion is not fun and I'm guessing it'll be some stupid interger overflow or dumb boundary condition hitting..."

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Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by albox＠free.fr 30 Jun '07

30 Jun '07

Could this issue be related to the other one with Medion laptops : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3160 Takashi, you said you were going to check the older patch that was working on ALSA 1.0.13, did you have a chance to do that ? Takashi Iwai wrote: > At Mon, 25 Jun 2007 15:16:41 -0500, > Matt Mullins wrote: >> On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: >>>> Yes, single_cmd mode is used always. >>> That's bad. >> I believe we have established that. What can I do about it? *Can* I >> do anything about it? What functions should I start debugging? Has >> this happened with other codecs before? What documentation will I >> need or want other than the Intel HDA and STAC9205 specifications? > > Similar problems have been seen in some laptops, too. I vaguely > remember that they need some ACPI-related boot options to fix it > somehow. I think it's likely irrelevant with the codec chip, but > rather a problem of the controller side. > >> I'm definitely willing to help work on this, as not many people have >> this hardware. I'm just new to the sound system, so I'm still trying >> to figure out what-calls-what. > > > Takashi > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

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Re: [alsa-devel] Creative Soundblaster X-Fi
by Ash Willis 30 Jun '07

30 Jun '07

Hi all, Sorry for straying a little off topic, but talk of these couple of chips got me thinking about unsupported hardware in general. Does anyone know the legal implications of reverse engineering with regards to producing new drivers? I myself wouldn't mind reversing a certain chip or two but I'm not too aware of the legality of it all, particularly in the case where the vendor is hostile or unwilling to release chip information. In general, is it legally acceptable to reverse engineer for the purposes of interoperability? Are vendors likely to make a noise if their hardware gains support under Linux without or against their consent? or is it more a matter of protecting the original product specification than hiding details of the product's interface? Takashi, may I ask your thoughts on this? What's your position on accepting reverse engineered code? (Let's assume that the driver works as well as one written in a more correct manner ;)) I'm fully aware that uncooperative vendors don't deserve to have their hardware supported, but I think that Linux users and Linux itself deserves the support. Ash -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains

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[alsa-devel] [1/2] 2.6.22-rc6: known regressions v2
by Michal Piotrowski 30 Jun '07

30 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc6. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions List of Aces Name Regressions fixed since 21-Jun-2007 Hugh Dickins 2 Andi Kleen 1 Björn Steinbrink 1 Bjorn Helgaas 1 Jean Delvare 1 Olaf Hering 1 Siddha, Suresh B 1 Trent Piepho 1 Ville Syrjälä 1 Unclassified Subject : Kernel hang on CMOS_READ References : http://lkml.org/lkml/2007/6/3/138 Submitter : Rodrigo Luiz <troops.of.doom(a)gmail.com> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown FS Subject : 2.6.22-rc4-git5 reiserfs: null ptr deref. References : http://lkml.org/lkml/2007/6/13/322 Submitter : Randy Dunlap <randy.dunlap(a)oracle.com> Handled-By : Vladimir V. Saveliev <vs(a)namesys.com> Status : problem is being debugged IDE Subject : 2.6.22-rcX: hda: lost interrupt References : http://lkml.org/lkml/2007/6/29/121 Submitter : David Chinner <dgc(a)sgi.com> Status : unknown SATA/PATA Subject : libata IT821X driver still fails! Hard-freezes system References : http://lkml.org/lkml/2007/6/14/82 Submitter : Rodney Gordon II <meff(a)spherevision.org> Status : unknown Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

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[alsa-devel] Creative Soundblaster X-Fi
by Tim Laszlo 29 Jun '07

29 Jun '07

Has anyone worked on developing an open source driver for the sound blaster x-fi cards? I've been sitting on one for about a year (mental note check support *before* buying..) and would love to use it already... Creative has promised drivers but they seem to be in no rush to support Linux. I'd like to try my hand at working on a driver for this, but without any technical specs from creative I wouldn't really know where to begin. Anyone have any suggestions on how to start? Anyone worked on this yet? Am I nuts for even considering this?? Timl

2 2

[alsa-devel] SALSA-Lib: Playback overrun on initial PCM start
by J. Scott Merritt 29 Jun '07

29 Jun '07

Dear List, Using SALSA-Lib 0.0.3, I call: snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0); snd_pcm_hw_params_malloc (&hw_params); snd_pcm_hw_params_any (playback_handle, hw_params); snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)); snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, 44100, 0); snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2); snd_pcm_hw_params_set_buffer_size_last (playback_handle, hw_params, &bfrsize); snd_pcm_hw_params_get_buffer_size (hw_params, &bfrsize); snd_pcm_hw_params (playback_handle, hw_params); snd_pcm_hw_params_free (hw_params); snd_pcm_sw_params_malloc (&sw_params); snd_pcm_sw_params_current (playback_handle, sw_params); snd_pcm_sw_params_set_avail_min (playback_handle, sw_params, 4096); snd_pcm_sw_params_set_start_threshold (playback_handle, sw_params, 10000U); snd_pcm_sw_params_set_xfer_align (playback_handle, sw_params, 1); snd_pcm_sw_params (playback_handle, sw_params); state = snd_pcm_state (playback_handle); frames_to_deliver = snd_pcm_avail_update (playback_handle); while ((frames_to_deliver = snd_pcm_avail_update (playback_handle)) > 4096) { snd_pcm_writei (playback_handle, &buf, 4096) } As soon as the buffer start threshold is reached, or alternatively if I manually start PCM stream, the PCM stream reports an overrun (i.e. "Broken Pipe"). If I then examine the PCM state, it is in the overrun (XRUN) state. The code is being cross-compiled and tested on an ARM processor (PXA270) which is running linux kernel 2.6.21. This same code sequence -does- operate properly (in the same ARM test environment) when compiled against AlsaLib 1.0.13. Another apparent (and perhaps relevent) difference between AlsaLib and SALSA-Lib is that if I open the PCM stream with O_NONBLOCK and then call snd_pcm_poll_descriptors, AlsaLib gives me a FD = 4, whereas SALSA-Lib reports an FD = 3; Thanks, Scott.

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Re: [alsa-devel] MOTU development
by Jonathan Woithe 29 Jun '07

29 Jun '07

Hi Per > I'd like to know more about the health on the connections to MOTU, if > they give any information about their products and how the development > has proceeded over the years. It kind of depends which MOTU devices you're asking about. There is, for example, an alsa driver for the MOTU Midi Timepiece AV (MTPAV) which presumedly makes the midi side of this available via the ALSA sequencer system. I can't speak for any of MOTU's PCI/PCIe cards. However, in connection with their more recent outboard interfaces MOTU have been, to put it mildly, extremely unhelpful. Some on the net go so far as to say they are hostile to Linux. The facts are that despite multiple people requesting information from MOTU on how to drive their firewire interfaces MOTU have responded every time with a very terse non-negotiable "no, never". I don't know if anyone's ever asked about their USB-equipped devices but I can't imagine the answer would be any different. Despite this, some progress is being made in supporting MOTU firewire devices under linux. For various reasons this support is not through ALSA but instead as part of the FFADO project. FFADO provides a userspace library for driving firewire audio devices in general and a JACK backend allowing such devices to be used by any jack-enabled software. An ALSA backend is planned but no work has been done towards this yet. Check out www.ffado.org for information on this project. Device support in FFADO varies. Some interfaces for which we have vendor support are doing very well, while supporting an interface such as the MOTUs without any documentation is challenging. However, the MOTU control protocol is pretty well understood at this point in time and it is possible to actually use the MOTU firewire devices for real work under Linux (as I have been for about 9 months now). The trap is that one needs to use an earlier revision of ffado at present. Back in February some key FFADO infrastructure changed. From before that until the last week or so I've been busy with production projects and so haven't had a chance to update the MOTU FFADO driver and restore its functionality. I'm back on the case now though and hope to make progress in the next month or so. If you want further details about all this please contact me off-list and I'll fill you in. Best regards jonathan

1 0

[alsa-devel] ALSA Application Programming Questions
by Paul Kavan 28 Jun '07

28 Jun '07

Hello all: I have been working on getting ALSA to work on my at91sam9260 with a simple voice codec for a few weeks. For the most part, I have my ARM board producing nice audio with aplay/arecord. Now I need a custom application that will allow me to create an 'intercom' like device with my ARM board. I am planning on writing the application as I do not think there is something out there that will accomplish what I want. I am really new to ALSA--as some of you know--and, as such, have never written an application for ALSA. I have found the ALSA driver api page on the alsa site. Wondering if there is any other good documentation on writing alsa applications? Any advice is greatly appreciated as well. One of my concerns is building the applications against the libs. I have the libs on my target machine. As my host machine has a version of my rfs on it, the libs are there as well. Just wondering how to accomplish cross-compiling new apps. I suppose I could copy the libs into the directory with my cross-compiling tools. Again, any advice would be great. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] sound on SMDK2410 working
by Jose Henrique Spahn Torres 28 Jun '07

28 Jun '07

Hi all, I was able to get sound on smdk2410 to work. I can play .wav files with the following asound.conf file: /etc $ more asound.conf pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } However the sound is distorted, in the sense that it sounds as played with a lower rate. I tried force a rate with -r 44100 or other rates, with no success. Can anyone point me to the right direction? Thanks, Henrique

1 0

[alsa-devel] SALSA-Lib: snd_pcm_hw_params_any clobbers memory
by J. Scott Merritt 28 Jun '07

28 Jun '07

Dear list, Using SALSA-Lib 0.0.2, I call snd_pcm_open snd_pcm_hw_params_malloc snd_pcm_hw_params_any The call to snd_pcm_hw_params_any appears to clobber memory beyond the end of the hw_params allocated by snd_pcm_hw_params_malloc (as reported by linking in -lmcheck). The code is being cross-compiled and tested on an ARM processor (PXA270) which is running linux kernel 2.6.21. This same code sequence -does- operate properly (in the same ARM test environment) when compiled against AlsaLib 1.0.13. I got "lost" somewhere around the hw_refine IOCTL .... Any suggestions appreciated :) Thanks, Scott.

2 2

[alsa-devel] MOTU development
by Per Eriksson 28 Jun '07

28 Jun '07

Dear list, My name is Per Eriksson, working as a musician with music and interactive content in Sweden. I'd like to know more about the health on the connections to MOTU, if they give any information about their products and how the development has proceeded over the years. I am thankful for any input. Best Regards Per Eriksson per(a)erinet.se

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#430833: libasound2: PCM plugin `lfloat' not built
by Elimar Riesebieter 28 Jun '07

28 Jun '07

forwarded 430833 alsa-devel(a)alsa-project.org -- On Wed, 27 Jun 2007 the mental interface of Mark Wooding told: > Package: libasound2 > Version: 1.0.14a-1 > Severity: normal > Tags: patch > > The PCM plugin `lfloat' isn't compiled into libasound. In particular, > this prevents the `plug' plugin from doing linear<->float conversion, > and in turn this means that hardly any ALSA clients can use the JACK > plugin from libasound2-plugins, because JACK uses FLOAT_LE format > throughout. > > The reason for this is quite simple: the configure script is broken! In > particular, the sense of the `softfloat' check used to decide whether to > disable the plugin is wrong! (That's aside from the question of whether > it's sensible to fail to build the thing just because it might be slow.) > We should solve this in general. It's not a Debian specific wish so forwarded upstream. > > Index: alsa-lib-1.0.14a/configure.in > =================================================================== > --- alsa-lib-1.0.14a.orig/configure.in 2007-06-27 16:24:58.000000000 > +0100 > +++ alsa-lib-1.0.14a/configure.in 2007-06-27 16:25:19.000000000 > +0100 > @@ -416,7 +416,7 @@ > build_pcm_share="no" > fi > > -if test "$softfloat" != "yes"; then > +if test "$softfloat" = "yes"; then > build_pcm_lfloat="no" > fi > > Index: alsa-lib-1.0.14a/configure > =================================================================== > --- alsa-lib-1.0.14a.orig/configure 2007-06-27 16:25:23.000000000 > +0100 > +++ alsa-lib-1.0.14a/configure 2007-06-27 16:25:38.000000000 +0100 > @@ -20946,7 +20946,7 @@ > build_pcm_share="no" > fi > > -if test "$softfloat" != "yes"; then > +if test "$softfloat" = "yes"; then > build_pcm_lfloat="no" > fi > > Elimar -- Experience is something you don't get until just after you need it!

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[alsa-devel] Sound disappears after a long run, pcm_rate plugin maybe
by Dave Airlie 28 Jun '07

28 Jun '07

Hi, I've been debugging this bug which I first noticed close to a year ago but worked around by disabling dmix. Now I've had to enable dmix and try and fix it. My system is SDL on alsa-lib-1.0.13 on 2.6.21 on intel 810 with a Realtek ALC655. I'm using SDL to open a 22050 16-bit stereo stream, and dmix is set to 48000. When I run sound in my single app (only one app running for testing purposes) around 20-24 hours later (I haven't got exact timings as I can't sit in front of the pc for so long) the sound stop s happening just in that app. I can run aplay separately and it plays fine. If I stop my app (ctrl-C and cont in gdb) or attach/detach strace, I get an XRUN and my SDL code calls snd_pcm_prepare and the sounds comes back. Now I can't tell you how much it sucks to only get a single debugging run every 24 hours for 1 break point and if gdb stuffs it up you lose big time. I think I've tracked it down to pcm_rate.c as in if I open the sound card at 48000 instead of 22050 it doesn't happen and I've added a lot of printfs to the rate code in 1.0.13 and gotten huge log file traces of it.. In a normal rate trace when things are working I get traces like this: snd_pcm_rate_sync_playback_area: 442023720 442023720 525 986185728 snd_pcm_rate_sync_playback_area: 442023720 442023720 1549 986185728 snd_pcm_rate_sync_playback_area: 442023720 442023720 1549 986185728 snd_pcm_rate_sync_playback_area: 442024660 442023720 1549 986185728 snd_pcm_rate_commit_area: 6 940 1024 1881 snd_pcm_rate_sync_playback_area: 940 442023720 1881 snd_pcm_rate_sync_playback_area: 442024660 442024660 525 986185728 snd_pcm_rate_sync_playback_area: 442024660 442024660 1549 986185728 snd_pcm_rate_sync_playback_area: 442024660 442024660 1549 986185728 snd_pcm_rate_sync_playback_area: 442025595 442024660 1549 986185728 snd_pcm_rate_sync_playback_area: 442025600 442024660 1549 986185728 snd_pcm_rate_commit_area: 946 940 1024 1881 snd_pcm_rate_sync_playback_area: 940 442024660 1881 snd_pcm_rate_sync_playback_area: 442025600 442025600 525 986185728 snd_pcm_rate_sync_playback_area: 442025600 442025600 1549 986185728 snd_pcm_rate_sync_playback_area: 442025600 442025600 1549 986185728 snd_pcm_rate_sync_playback_area: 442026540 442025600 1549 986185728 snd_pcm_rate_commit_area: 5 940 1024 1881 snd_pcm_rate_sync_playback_area: 940 442025600 1881 snd_pcm_rate_sync_playback_area: 442026540 442026540 525 986185728 snd_pcm_rate_sync_playback_area: 442026540 442026540 1549 986185728 snd_pcm_rate_sync_playback_area: 442026540 442026540 1549 986185728 snd_pcm_rate_sync_playback_area: 442027476 442026540 1549 986185728 snd_pcm_rate_sync_playback_area: 442027480 442026540 1549 986185728 playback are debug code slave_size = snd_pcm_avail_update(slave); if (slave_size < 0) return slave_size; fprintf(stderr,"%s: %u %d %u %u\n", __func__, appl_ptr, rate->last_commit_ptr, slave_size, pcm->boundary); if (appl_ptr < rate->last_commit_ptr) xfer = appl_ptr - rate->last_commit_ptr + pcm->boundary; else xfer = appl_ptr - rate->last_commit_ptr; while (xfer >= pcm->period_size && (snd_pcm_uframes_t)slave_size >= rate->gen.slave->period_size) { err = snd_pcm_rate_commit_next_period(pcm, rate->last_commit_ptr % pcm->buffer_size); fprintf(stderr, "%s: %u %u %u\n", __func__, xfer, rate->last_commit_ptr, pcm->buffer_size); if (err == 0) commit area debug fprintf(stderr,"%s: %u %u %u %u\n", __func__, appl_offset, size, slave_size, pcm->buffer_size); areas = snd_pcm_mmap_areas(pcm); if (cont >= size) { In the not so good case (with different debug - two sites and I hadn't access to my debug code from one so I wrote it again) you get large numbers of snd_pcm_rate_sync_playback_area: 152605942 912 152385750 snd_pcm_rate_sync_playback_area: 152605942 912 152385750 snd_pcm_rate_sync_playback_area: 152606412 912 152385750 snd_pcm_rate_sync_playback_area: 152606412 912 152385750 snd_pcm_rate_sync_playback_area: 152606705 912 152385750 snd_pcm_rate_sync_playback_area: 152606882 912 152385750 snd_pcm_rate_sync_playback_area: 152606882 912 152385750 snd_pcm_rate_sync_playback_area: 152607352 912 152385750 snd_pcm_rate_sync_playback_area: 152607352 912 152385750 ...... snd_pcm_rate_sync_playback_area: 152769972 912 152385750 snd_pcm_rate_sync_playback_area: 152770381 1936 152385750 snd_pcm_rate_commit_area: 1411 572 470 snd_pcm_rate_sync_playback_area: 384631 1 1936 152385750 739770368 snd_pcm_rate_sync_playback_area: 152770442 912 152386220 .... more of these.... snd_pcm_rate_sync_playback_area: 152910502 912 152386220 snd_pcm_rate_sync_playback_area: 152910502 912 152386220 snd_pcm_rate_sync_playback_area: 152910972 912 152386220 snd_pcm_rate_sync_playback_area: 152910972 912 152386220 appl ptr is 152910972 recalc: 3022344724 1411 3072 470 1024 snd_pcm_rate_sync_playback_area: 152910972 1936 152386220 snd_pcm_rate_commit_area: 1411 1042 470 snd_pcm_rate_sync_playback_area: 524752 1 1936 152386220 739770368 snd_pcm_rate_sync_playback_area: 152910972 1936 152386690 snd_pcm_rate_commit_area: 1411 101 470 snd_pcm_rate_sync_playback_area: 524282 1 1936 152386690 739770368 snd_pcm_rate_sync_playback_area: 152910972 1936 152387160 ... this pattern repeats one commit per two playbacks until it goes back in the top... the debug code for these are recalc: fprintf(stderr,"%s: %lu %lu %lu %lu %lu\n", __func__, val, pcm->buffer_size, slave->buffer_size, pcm->period_size, slave->period_size); commit area areas = snd_pcm_mmap_areas(pcm); fprintf(stderr,"%s: %u %u %u\n", __func__, pcm->buffer_size, appl_offset, size); if (cont >= size) { playback area: fprintf(stderr,"%s: %u %d %d\n", __func__, appl_ptr, slave_size, rate->last_commit_ptr); if (appl_ptr < rate->last_commit_ptr) xfer = appl_ptr - rate->last_commit_ptr + pcm->boundary; else xfer = appl_ptr - rate->last_commit_ptr; while (xfer >= pcm->period_size && (snd_pcm_uframes_t)slave_size >= rate->gen.slave->period_size) { err = snd_pcm_rate_commit_next_period(pcm, rate->last_commit_ptr % pcm->buffer_size); fprintf(stderr,"%s: %u %d %d %u %u\n", __func__, xfer, err, slave_size, rate->last_commit_ptr, pcm->boundary); I'm hoping it is just some calculation in the rate plugin overflowing and I was blaming SDL but I've fixed up the codepaths to look like aplay and it still happened.. any hints would be appreciated on a) what that rate code is actually doing in the fail case, b) what the rate code is doing in the non-fail case :-) Dave.

2 1

[alsa-devel] SoC multiple streams
by Steve Longerbeam 28 Jun '07

28 Jun '07

Hi all, I'm developing an SoC driver for a sound engine that supports playing up to 64 simultaneous substreams, all mixed in h/w to a stereo DAC. But it doesn't appear that the SoC framework supports multiple substreams, according to soc_new_pcm(). Am I missing something, or is this on the asoc development roadmap? Thanks, Steve

3 3

[alsa-devel] SoC Machine Code Suggestions
by Paul Kavan 27 Jun '07

27 Jun '07

First, thanks to all for helping me get the at91 soc stuff working with my board and codec. It now plays telecom quality audio using aplay and records it as well using arecord. For those who were following my trials and did not hear what the solution was, essentially, when the at91-ssc code goes into playback mode, it only activates the TX side of the SSC (PCM) bus. Similarly, when it is in capture mode, it only activates the RX side of the bus. This certainly makes sense and works in most situations I am sure. However, I needed both the RX and TX frames operating in both playback and capture. As such, I had to modify at91-ssc slightly to accomplish this. Now that I have decent audio, I want to clean up around the edges of the audio and determine the best way to proceed with my application code. First. when the pcm interface starts/stops, I get a pop that comes from the codec analog output transitioning some. Now, there may be some hardware fixes to this, but what I would like to do it mute the amp until the last possible second before audio is played. I can accomplish this with a simple gpio call, but am not sure where best to turn the mute on and off. Any suggestions with that would be appreciated. Also, my next phase of development is to determine whether I need to write my own application to accomplish what I want to do, or if there is an application out there that will do the trick--and that I can cross-compile to an ARM. What I would like is an application I will stream audio (half-duplex) between two ARM boards connected via ethernet. I would need to be able to control the start/stop of the audio. Sometimes this control will depend on the state of the machine and other times it will be activate with a button press by the user. I hope that makes sense. Obviously it would have to seamlessly stream to the codec via ALSA. Also, there may be times when it streams to a pc. Now, the codec uses 8-bit ulaw, so this code if it were on a pc, would need to be able to play that on the pc in its native state or some type of conversion on the fly. I guess you can think of the application as similar to voip, but half duplex instead of full duplex. Does anyone know if something like this exists that I might use. If not, how hard would it be to code this for ALSA? Thanks again and I look forward to your responses. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 2

Re: [alsa-devel] SALSA-Lib version 0.0.1 release
by Takashi Iwai 27 Jun '07

27 Jun '07

At Mon, 25 Jun 2007 23:32:39 +0100, Adrian McMenamin wrote: > > > You can check ldd to the app which shlib it's linked with. > > It should be linked with libsalsa, not libasound. > > > > > > Takashi > > > > I expect I am being a bit thick here but what app? I use mpg123 and > it's not linked with any alsa libs. > > / # ldd ./mpg123 > libm.so.0 => /lib/libm.so.0 (0x2956d000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x2958b000) > libc.so.0 => /lib/libc.so.0 (0x295aa000) > ld-uClibc.so.0 => /lib/ld-uClibc.so.0 (0x29558000) mpg123 is an OSS app. > / # ldd aplay > libasound.so.2 => /usr/lib/libasound.so.2 (0x2956d000) > libm.so.0 => /lib/libm.so.0 (0x29652000) > libdl.so.0 => /lib/libdl.so.0 (0x29670000) > libpthread.so.0 => /lib/libpthread.so.0 (0x29683000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x296a4000) > libc.so.0 => /lib/libc.so.0 (0x296c3000) > ld-uClibc.so.0 => /lib/ld-uClibc.so.0 (0x29558000) It's liekly linked with the old libasound. SALSA-lib is a source-level compatible library and it's _not_ binary-level compatible. Thus, it's no drop-in replacement. After installing salsa-lib (better use --enable-everything option to make a libasound symlink), then you'll have to rebuild the other apps linked with alsa-lib again. Takashi

1 0

Re: [alsa-devel] ALSA always writes _whole_ periods, right?
by Takashi Iwai 27 Jun '07

27 Jun '07

At Sun, 24 Jun 2007 11:34:56 +0200, Joachim Förster wrote: > > Hi Takashi, > > thanks for your answer. > > On Wed, 2007-06-20 at 12:45 +0200, Takashi Iwai wrote: > > > Ok, ahm, I think my question was not about what I really wanted to know > > > - sorry. I have to reformulate it: In fact I would like to know, how the > > > ALSA-library mmap code behaves regarding the last chunk of samples, for > > > example: Consider aplay, playing a wav file and the wav file's > > > size/number of samples does _not_ fit to a multiple of a period's size. > > > Then, does the mmap code copy these last samples as is _or_ does it copy > > > one whole last period (which is filled with zeros)? > > > > No, you have to fill silence by yourself before committing the mmap > > chunk. The mmap code itself doesn't care about it. > > By "fill silence by yourself", you mean the application (user-space) > which is using ALSA library (with mmap), right? So this is not in the > responsibility of the sound driver, right? Right. The driver has some functionality to fill silence automatically, but it's better to avoid to play with it :) > Hmmm, so if an application does _not_ transfer sound data in a multiple > of period size, the _last_ "period" will be "incomplete". Then, my > question is: Does ALSA expect to receive a call of snd_period_elapsed() > after these last samples are played, although the it was _not_ a > complete period? I think it might depend on the hardware. Some hardwares are available to stop at the exact position, but most hardware are not. So, it's _safer_ to send always the data in the period size. And, yes, for the last period with less data, the app should fill the data to align to the period size. Takashi

1 0

[alsa-devel] Getting physical address of the DMA buffer
by Pharaoh . 27 Jun '07

27 Jun '07

Hi I have allocated the DMA buffer using snd_pcm_lib_malloc_pages() and it was preallocated as: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data (GFP_KERNEL), 128 * 1024, 128 * 1024); and then, snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); None of these calls fail. The physical address of the allocated buffer is shown as 0 in my case since according to 'Writing an ALSA driver' the physical address of the DMA buffer will not be accessble from the main CPU. Is that why the physical address is shown as 0? Also, in my pointer function, I need the DMA buffers physical address to calculate the position in buffer etc. How can do that since runtime->dma_addr is 0 for me? Apologies for longish mail but I have to paste the actual paragraphs here to clear my understanding in less number of emails by avoiding further emails. In chapter 5, in DMA buffer information, it is said that, 1. The dma_area holds the buffer pointer (the logical address). You can call memcpy from/to this pointer. Meanwhile, dma_addr holds the physical address of the buffer. This ﬁeld is speciﬁed only when the buffer is a linear buffer. and in descrition of prepare callback it says, 2. The physical address of the allocated buffer is set to runtime->dma_area. Statements 1 and 2 contradict each other or I am mis reading them? How should I get the physical address in pointer callbacks? As runtime->dma_area or runtime->dma_addr which is 0? -pharaoh.

2 4

[alsa-devel] Input/output error while playing a file using aplay
by Pharaoh . 26 Jun '07

26 Jun '07

Hi I am testing my audio driver using aplay. I trying to play an audio file using: #aplay -Dplughw:0,0 --rate=44100 --format=S16 --channels=2 test.wav The log is as follows, snd_card_omap_alsa_open start snd_card_omap_alsa_open end Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo snd_omap_alsa_hw_params start snd_omap_alsa_prepare start snd_omap_alsa_prepare end snd_omap_alsa_trigger start audio_start_dma_chain start audio_start_dma_chain end omap_start_alsa_sound_dma end snd_omap_alsa_trigger end snd_omap_alsa_pointer start audio_get_dma_pos start audio_get_dma_pos end It waits here for sometime and then I get snd_omap_alsa_trigger starte error: Input/output error Then the things are wound up as: audio_stop_dma start omap_stop_alsa_sound_dma start omap_stop_alsa_sound_dma end omap_clear_alsa_sound_dma start omap_clear_alsa_sound_dma end snd_omap_alsa_trigger end snd_omap_alsa_hw_free start snd_omap_alsa_hw_free end snd_card_omap_alsa_close start snd_card_omap_alsa_close end So basically the problem is in trigger where it waits for something and then winds up things. Any idea, what could be the reason. I am thinking it has to do something with DMA interrupts etc.

2 14

[alsa-devel] at91 SoC Modifications
by Paul Kavan 26 Jun '07

26 Jun '07

Hello: Okay, so after probing the various pcm lines, I think I see that the format used for the pcm is a little bit off of what my codec likes to play nice. In my machine/codec driver, I see that I have some lines: /* set cpu DAI configuration */ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); I believe this is where it is selecting the format. In at91-ssc.c, I see that this format is given as: case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: /* * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. * * The SSC transmit and receive clocks are generated from the * MCK divider, and the BCLK signal is output on the SSC TK line. */ rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) | (( 1 << 16) & AT91_SSC_STTDLY) | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) | (( 0 << 16) & AT91_SSC_FSLEN) | (((channels - 1) << 8) & AT91_SSC_DATNB) | (( 1 << 7) & AT91_SSC_MSBF) | (( 0 << 5) & AT91_SSC_LOOP) | (((bits - 1) << 0) & AT91_SSC_DATALEN); tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) | (( 1 << 16) & AT91_SSC_STTDLY) | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) | (( 0 << 23) & AT91_SSC_FSDEN) | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) | (( 0 << 16) & AT91_SSC_FSLEN) | (((channels - 1) << 8) & AT91_SSC_DATNB) | (( 1 << 7) & AT91_SSC_MSBF) | (( 0 << 5) & AT91_SSC_DATDEF) | (((bits - 1) << 0) & AT91_SSC_DATALEN); break; I should have caught this before--I spent enough time dealing with the HW registers. The simple ssc pcm driver I wrote sets the regs a little differently to get the peak performance out of the codec, e.g. In RCMR, I have STTDLY=0, AT91_SSC_CKI_FALLING, AT91_SSC_CKS_PIN. In RFMR everything looks okay except that I have FSOS_NONE as I borrow it from TF0 In TCMR the STTDLY=0 in mine In TFMR I do nothing with the FSEDGE and FSDEN and I have FSLEN=7 and DATADEF=1. So, I left with a decision. Do I add a mode or do I modify DSP A to fit my needs? I am not sure how much coding it would take to add a mode--I *do* know it would take little time to modify DSP A, but I am reluctant to make a change to something that is probably 'standard'. Any thoughts on this? Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] Fwd: Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Matt Mullins 25 Jun '07

25 Jun '07

On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > Yes, single_cmd mode is used always. > That's bad. I believe we have established that. What can I do about it? *Can* I do anything about it? What functions should I start debugging? Has this happened with other codecs before? What documentation will I need or want other than the Intel HDA and STAC9205 specifications? I'm definitely willing to help work on this, as not many people have this hardware. I'm just new to the sound system, so I'm still trying to figure out what-calls-what. - Matt

1 0

[alsa-devel] Intel Realtek ALC268 don't sound
by antoniorc＠4lineas.net 25 Jun '07

25 Jun '07

Hi, I tried latest HG driver version of Alsa, with patch included for ALC268. Apparently card is detected and mixer show correctly volumes bars, but don't sound. I tried option 3stack and auto in /etc/modprobe.d/alsa-base but nothing difference. I've a HP Pavilion dv6528 with Santa Rosa. Please tell me that I can do? Thanks

1 0

[alsa-devel] problem with audigy2 ZS remote
by Stefano Radaelli 25 Jun '07

25 Jun '07

Hi all! Sorry for my bad english. I have a problem with the livedrive of an audigy2 ZS. After some upgrades (on a gentoo linux system) the eternal panel seems to stop working. Even after the "initialization" (with echo on the midi device), the IR led not blink on remote action. This worked correctly in the past but after some upgrade (i don't know exactly what) it stoipped working. I don't known where i have to look (logs?) or what type of debugging it's necessary. Help me please

1 0

[alsa-devel] ARM in embedded audio
by Alan Horstmann 25 Jun '07

25 Jun '07

Hi, Can anyone help point me to info on using ARM in embedded audio systems? Does anyone use a low-cost chip with I2S interface? Thanks Alan

3 5

Re: [alsa-devel] [alsa-cvslog] alsa-lib: Add struct timeval and timespec definition when _POSIX_C_SOURCE is not defined
by Sergey Vlasov 25 Jun '07

25 Jun '07

On Fri, Oct 06, 2006 at 11:01:44AM +0200, Jaroslav Kysela wrote: > changeset: 2218:603411fd0c77712cab43cdd5afcf0ff4ebe51a7f > tag: tip > user: perex > date: Fri Oct 06 11:01:40 2006 +0200 > files: include/global.h > description: > Add struct timeval and timespec definition when _POSIX_C_SOURCE is not defined > > FIXME: It might cause problems on some platforms when tv_usec is not long type. And now it also breaks compilation of the following code (at least with glibc 2.5): #include <sys/select.h> #include <alsa/global.h> $ gcc -c --std=c99 x.c In file included from x.c:2: /usr/include/alsa/global.h:134: error: redefinition of 'struct timeval' /usr/include/alsa/global.h:139: error: redefinition of 'struct timespec' (Without #include <sys/select.h> only 'struct timespec' is redefined.) IMHO attempts to define structures which should be defined by system headers are too dangerous - if someone is using preprocessor flags which hide required parts of system headers, it is their problem, and the risk of mismatch is too high. BTW, <time.h> does not provide 'struct timeval' at all, even with _GNU_SOURCE - it defines only 'struct timespec'; however, this does not cause problems, because ALSA headers currently use only pointers to snd_timestamp_t, which works even without a complete definition of 'struct timeval'. We can try to include <sys/time.h> to get 'struct timeval', or even <sys/select.h> (which currently seems to provide both 'struct timeval' and 'struct timespec' even with -ansi, and http://www.opengroup.org/onlinepubs/009695399/basedefs/sys/select.h.html tells that it should always provide their definitions). > > > diff -r 7bc021fb4bf1027b23aa5429ceda1e1e35c7d126 -r 603411fd0c77712cab43cdd5afcf0ff4ebe51a7f include/global.h > --- a/include/global.h Fri Oct 06 10:38:39 2006 +0200 > +++ b/include/global.h Fri Oct 06 11:01:40 2006 +0200 > @@ -130,6 +130,18 @@ int snd_shm_area_destroy(struct snd_shm_ > > int snd_user_file(const char *file, char **result); > > +#ifndef _POSIC_C_SOURCE > +struct timeval { > + time_t tv_sec; /* seconds */ > + long tv_usec; /* microseconds */ > +}; > + > +struct timespec { > + time_t tv_sec; /* seconds */ > + long tv_nsec; /* nanoseconds */ > +}; > +#endif > + > /** Timestamp */ > typedef struct timeval snd_timestamp_t; > /** Hi-res timestamp */

2 3

[alsa-devel] SALSA library in ARM platform
by Nobin Mathew 25 Jun '07

25 Jun '07

Hi, I tried compiling SALSA for ARM. I got few errors. CC=arm-none-linux-gnueabi-gcc ./configure --target=arm-linux --host=i386 --prefix=/usr make Then I am getting following errors See the bottom part /bin/bash ../libtool --tag=CC --mode=compile arm-none-linux-gnueabi-gcc -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE=\"salsa-lib\" -DVERSION=\"0.0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DSTDC_HEADERS=1 -I. -Wall -g -g -O2 -MT mixer.lo -MD -MP -MF .deps/mixer.Tpo -c -o mixer.lo mixer.c arm-none-linux-gnueabi-gcc -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE=\"salsa-lib\" -DVERSION=\"0.0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DSTDC_HEADERS=1 -I. -Wall -g -g -O2 -MT mixer.lo -MD -MP -MF .deps/mixer.Tpo -c mixer.c -o mixer.o In file included from hcontrol.h:36, from mixer.h:25, from mixer.c:30: hctl_macros.h: In function 'snd_hctl_elem_tlv_read': hctl_macros.h:136: warning: 'snd_ctl_elem_tlv_read' is deprecated (declared at ctl_macros.h:1131) hctl_macros.h: In function 'snd_hctl_elem_tlv_write': hctl_macros.h:142: warning: 'snd_ctl_elem_tlv_write' is deprecated (declared at ctl_macros.h:1138) hctl_macros.h: In function 'snd_hctl_elem_tlv_command': hctl_macros.h:148: warning: 'snd_ctl_elem_tlv_command' is deprecated (declared at ctl_macros.h:1145) mv -f .deps/mixer.Tpo .deps/mixer.Plo /bin/bash ../libtool --tag=CC --mode=link arm-none-linux-gnueabi-gcc -Wall -g -g -O2 -version-number 0:0:1 -o libsalsa.la -rpath /usr/lib control.lo cards.lo pcm.lo pcm_params.lo pcm_misc.lo hcontrol.lo mixer.lo mkdir .libs libtool: link: CURRENT `' must be a nonnegative integer libtool: link: `0:0:1' is not valid version information make[1]: *** [libsalsa.la] Error 1 make[1]: Leaving directory `/home/nmathew/Desktop/salsa-lib-0.0.1/src' make: *** [all-recursive] Error 1

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[alsa-devel] [PATCH] hda: support for S/PDIF out on ASUS M2V
by Mike Crash 25 Jun '07

25 Jun '07

Added support for S/PDIF digital output on ASUS M2V motheboard - added new model "3stack-660-digout" and ALC660VD_3ST_DIG Apply to 1.0.14 driver file patch_realtek.c Signed-off-by: Mike Crash <mike(a)mikecrash.com> *** patch_realtek.c 2007-06-04 09:28:54.000000000 +0200 --- patch_realtek2.c 2007-06-16 21:46:14.000000000 +0200 *************** enum { *** 117,118 **** --- 117,119 ---- ALC660VD_3ST, + ALC660VD_3ST_DIG, ALC861VD_3ST, *************** static const char *alc861vd_models[ALC86 *** 9466,9467 **** --- 9467,9469 ---- [ALC660VD_3ST] = "3stack-660", + [ALC660VD_3ST_DIG]= "3stack-660-digout", [ALC861VD_3ST] = "3stack", *************** static struct snd_pci_quirk alc861vd_cfg *** 9477,9479 **** SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), ! SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), --- 9479,9481 ---- SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), ! SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), *************** static struct alc_config_preset alc861vd *** 9501,9502 **** --- 9503,9517 ---- }, + [ALC660VD_3ST_DIG] = { + .mixers = { alc861vd_3st_mixer }, + .init_verbs = { alc861vd_volume_init_verbs, + alc861vd_3stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), + .dac_nids = alc660vd_dac_nids, + .dig_out_nid = ALC861VD_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .adc_nids = alc861vd_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), + .channel_mode = alc861vd_3stack_2ch_modes, + .input_mux = &alc861vd_capture_source, + }, [ALC861VD_3ST] = {

2 1

[alsa-devel] [PATCH] MPU401 lockups
by Serge Nikolaenko 25 Jun '07

25 Jun '07

Hello, The attachement is the fix for mpu to prevent lockups/freezes. Added mutexes with down_trylock() calls, to prevent blocking. Midi in/out not tested - I don't have no midi devices at the moment. My soundcard is ESI Juli@. MB integrated midi is disabled in bios. Last system messages after system freeze when running wine/jackd: kernel: cmd: 0xff failed at 0xac0c (status = 0xb, data = 0x0) ... kernel: BUG: soft lockup detected on CPU#0! kernel: kernel: Call Trace: kernel: <IRQ> [<ffffffff802b0bdb>] softlockup_tick+0xdb/0xf6 kernel: [<ffffffff8028f5d0>] update_process_times+0x42/0x68 kernel: [<ffffffff80271f0c>] smp_local_timer_interrupt+0x34/0x55 kernel: [<ffffffff802725e8>] smp_apic_timer_interrupt+0x51/0x69 kernel: [<ffffffff8025ace6>] apic_timer_interrupt+0x66/0x70 kernel: [<ffffffff889c600d>] :snd_mpu401_uart:mpu401_read_port+0x4/0x8 kernel: [<ffffffff889c62f0>] :snd_mpu401_uart:_snd_mpu401_uart_interrupt+0x4c/0x70 kernel: [<ffffffff889c6391>] :snd_mpu401_uart:snd_mpu401_uart_interrupt+0x13/0x1a kernel: [<ffffffff88a1f34b>] :snd_ice1724:snd_vt1724_interrupt+0x4e/0x13a kernel: [<ffffffff80210b93>] handle_IRQ_event+0x25/0x53 kernel: [<ffffffff802b202a>] handle_fasteoi_irq+0x92/0xd1 kernel: [<ffffffff8026862a>] do_IRQ+0xff/0x16d kernel: [<ffffffff8025a631>] ret_from_intr+0x0/0xa kernel: <EOI> [<ffffffff802f5df7>] proc_info_read+0x0/0xb9 kernel: [<ffffffff802f5df9>] proc_info_read+0x2/0xb9 kernel: [<ffffffff8020b14e>] vfs_read+0xcb/0x173 kernel: [<ffffffff80211847>] sys_read+0x45/0x6e kernel: [<ffffffff8025a11e>] system_call+0x7e/0x83

2 1

Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Takashi Iwai 25 Jun '07

25 Jun '07

[Please don't strip Cc to ML!] At Tue, 19 Jun 2007 15:54:42 -0500, Matt Mullins wrote: > > I played around with probe_mask, and found that probe_mask=0x1 causes > the error, but probe_mask=0x2 does not cause the timeout. Codec 1 is > my Sigmatel STAC9205, which is what plays sound on my laptop, and > codec 2 is my Conexant modem. The modem isn't the codec causing > problems; the audio codec is! OK, good to know. Does this problem occur with S2D? What is the very first command that triggers this timeout error? I'm wondering what is the difference between them -- according to your report, reloading the module fixes the problem. But, the resume callback is almost identical with the re-initiailzation. Perhaps it's something to do with the power-state management. What about the patch below? Takashi diff -r c531d63e820b sound/pci/hda/hda_intel.c --- a/sound/pci/hda/hda_intel.c Wed Jun 20 12:03:09 2007 +0200 +++ b/sound/pci/hda/hda_intel.c Wed Jun 20 12:09:40 2007 +0200 @@ -1450,7 +1450,7 @@ static int azx_suspend(struct pci_dev *p pci_disable_msi(chip->pci); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + /* pci_set_power_state(pci, pci_choose_state(pci, state)); */ return 0; }

2 4

[alsa-devel] hda-intel without headphone output
by Robert Lemmen 25 Jun '07

25 Jun '07

hi everyone, i have a fujitsu siemens amilo pro v3515 notebook, with a via VT8251/ VT8237A chipset. sound works fine through the speakers, but i don't get anything through the headphones, no headphone channel shows up in the mixer, and the headphone jack switch doesn't mute the speakers. i have alsa 1.0.13, and a 2.6.18 kernel. i tried all "model" settings that i could find in ALSA-Configuration in the kernel tree. lspci -nv gives 04:01.0 0403: 1106:3288 (rev 10) Subsystem: 1734:10cb Flags: bus master, fast devsel, latency 0, IRQ 58 Memory at c9100000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 (only the stanza for the sound card, yell if you need more), and /proc/asound/card0/codec#0 gives: Codec: Generic 14f1 ID 5045 Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x173410cb Revision Id: 0x100100 Default PCM: rates 0x140, bits 0x0e, types 0x1 Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x28 0x28] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x19* 0x17 Node 0x11 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x19* 0x17 Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0xab 0xab] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x20: IN Power: 0x0 Connection: 2 0x19* 0x17 Node 0x13 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x214571f0: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Yellow Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a7012e: [Fixed] Mic at Int Riser Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0b, mute=1 Amp-Out vals: [0x06] Node 0x17 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x14, nsteps=0x2b, stepsize=0x05, mute=1 Amp-In vals: [0x94 0x94] [0x17 0x17] [0x94 0x94] [0x94 0x94] [0x94 0x94] Power: 0x0 Connection: 5 0x19 0x14 0x12 0x11 0x15 Node 0x18 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates 0x040, bits 0x06, types 0x5 Node 0x19 [Audio Output] wcaps 0xc11: Stereo PCM: rates 0x540, bits 0x0e, types 0x1 Power: 0x0 Node 0x1a [Audio Input] wcaps 0x100d0b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x17, stepsize=0x05, mute=1 Amp-In vals: [0x17 0x17] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Power: 0x0 Connection: 5 0x17 0x14 0x12 0x11 0x15* Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono please tell me if there is anything i can do, any additional information you need, or any general tipps. thanks robert -- Robert Lemmen http://www.semistable.com

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[alsa-devel] [1/2] 2.6.22-rc6: known regressions
by Michal Piotrowski 25 Jun '07

25 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc6. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions *STATISTICS* (a.k.a. list of aces) Name Regressions fixed since 21-Jun-2007 Andi Kleen 1 Hugh Dickins 1 Jean Delvare 1 Unclassified Subject : Kernel hang on CMOS_READ References : http://lkml.org/lkml/2007/6/3/138 Submitter : Rodrigo Luiz <troops.of.doom(a)gmail.com> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown Memory management Subject : 2.6.22-rc5-yesterdaygit with VM debug: BUG in mm/rmap.c:66: anon_vma_link ? References : http://lkml.org/lkml/2007/6/25/45 Submitter : Petr Vandrovec <petr(a)vmware.com> Status : unknown SATA/PATA Subject : libata IT821X driver still fails! Hard-freezes system References : http://lkml.org/lkml/2007/6/14/82 Submitter : Rodney Gordon II <meff(a)spherevision.org> Status : unknown Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

1 0

[alsa-devel] alsa-lib: pcm_softvol.c: attenuation curve - it seems to be too quiet at full volume
by John L. Utz III 25 Jun '07

25 Jun '07

Hello All; I am in release testing with a fix to our product to use 4 channels on a CMEDIA 8738-MC6 (previously we only used 2 channels on the CMEDIA and a separate soundcard for the 2 back channels). The 8738 is a pretty funny old duck of a chip and i am trying to work around a subset of it's many idiosyncracies. :-) The most glaring defect is the absence of hard volume on the back channels *and* that the back channel is louder than the front channel. Obviously, this is a great candidate for softvol! Softvol works quite well *except* that the max volume for the rear channel is now much quieter than it is without softvol. :-( Is there anything i can do to alter this behavior? I notice that there is an initial minimal DB in the file: #define PRESET_MIN_DB -51.0 is this where everything starts from? if i bump this up, will i get a higher max volume? tnx! johnu

2 1

[alsa-devel] Query regarding pointer callback
by Pharaoh . 25 Jun '07

25 Jun '07

Hi In Writing an ALSA driver, it is said that pointer callback is called usually from the buffer-update routine in the pcm middle layer, which is invoked when snd_pcm_period_elapsed() is called in the interrupt routine. Then the pcm middle layer updates the position and calculates the available space, and wakes up the sleeping poll threads, etc What are the other places from where pointer callback is called? If it is not called from any other routine then does it mean that interrupts are occuring?

2 2

[alsa-devel] Intel Realtek ALC268 don't sound
by Antonio Rguez. Capita 25 Jun '07

25 Jun '07

Hi, I tried lastest HG driver version of Alsa, with patch included for ALC268. Apparently card is detected and mixer show correctly volumes bars, but don't sound. I tried option 3stack and auto in /etc/modprobe.d/alsa-base but nothing difference. I've a HP Pavilion dv6528 with Santa Rosa. Please tell me that I can do? Thanks

1 0

Re: [alsa-devel] No sound on Medion laptop (intel 82801G)
by albox＠free.fr 25 Jun '07

25 Jun '07

Here is the result : $ sudo dmesg -c $ sudo rmmod snd-hda-intel && sudo modprobe snd-hda-intel probe_mask=1 $ dmesg [ 638.324000] ACPI: PCI interrupt for device 0000:00:1b.0 disabled [ 638.368000] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 [ 638.368000] PCI: Setting latency timer of device 0000:00:1b.0 to 64 [ 638.400000] ALSA /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0x3 [ 638.576000] ALSA /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1820: hda_codec: model 'medion' is selected for config 161f:2054 (Medion laptop) I can hear a quick sound (like a "bip") when I enter the command but after that, the sound doesn't work.

1 0

[alsa-devel] No sound on Medion laptop (intel 82801G)
by albox＠free.fr 25 Jun '07

25 Jun '07

Hi, I reported a bug about medion laptops having an intel 82801G sound card that doesn't work with ALSA 1.0.14rc4 on Kubuntu : https://bugtrack.alsa-project.org/alsa-bug/bug_view_advanced_page.php?bug_i… I didn't get any news yet. As I said, I'm willing to help. Any ideas about what I should check ? Is there any web page that could help me to detect the problem ? Thanks

2 1

[alsa-devel] $1000 bounty for getting audio working under linux on a Osprey 440
by Nathan Allen Stratton 25 Jun '07

25 Jun '07

Need to get audio working on Osprey 440 card. http://www.viewcast.com/pages.asp?page=osprey-440 The card is bt878 based, and ViewCast says it is supported on linux however not something they support. Audio is not detected by default, but does how up when you do a modprobe snd-bt87x load-all(see below). However I have not been able to pull audio from the card (only get high pitched noise). Bid may require patching bt878.c if modprobe options don't work. Only mono support is required. [root@video share]# amixer Simple mixer control 'Bass',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 48 Mono: 30 [62%] Simple mixer control 'Bass Boost',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Treble',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 48 Mono: 24 [50%] Simple mixer control 'PCM',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 44 Capture 0 - 15 Mono: Capture 15 [100%] [15.00dB] [on] Front Left: Playback 44 [100%] [3.00dB] [on] Front Right: Playback 44 [100%] [3.00dB] [on] Simple mixer control 'PCM Capture Source',0 Capabilities: enum Items: 'PCM' 'Mic' 'Input 2' Item0: 'Input 2' Simple mixer control 'PCM',1 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 656 Mono: Capture 0 [0%] [-41.00dB] Simple mixer control 'Mic',0 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 0 [0%] [16.00dB] Simple mixer control 'Auto Gain Control',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] [root@video share]# amixer -c 1 Simple mixer control 'FM',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Mic/Line',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] Simple mixer control 'Capture',0 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 15 [100%] Simple mixer control 'Capture Boost',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'TV Tuner',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] :::::::::::::: cards :::::::::::::: 0 [INport ]: USB-Audio - INport Xitel INport at usb-0000:00:1d.0-1, full speed 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8001000, irq 16 2 [Bt878_1 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8003000, irq 17 3 [Bt878_2 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8005000, irq 18 4 [Bt878_3 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8007000, irq 19 :::::::::::::: devices :::::::::::::: 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control 7: [ 1- 1]: digital audio capture 8: [ 1- 0]: digital audio capture 9: [ 1] : control 10: [ 2- 1]: digital audio capture 11: [ 2- 0]: digital audio capture 12: [ 2] : control 13: [ 3- 1]: digital audio capture 14: [ 3- 0]: digital audio capture 15: [ 3] : control 16: [ 4- 1]: digital audio capture 17: [ 4- 0]: digital audio capture 18: [ 4] : control :::::::::::::: hwdep :::::::::::::: *** INport: directory *** :::::::::::::: modules :::::::::::::: 0 snd_usb_audio 1 snd_bt87x 2 snd_bt87x 3 snd_bt87x 4 snd_bt87x *** oss: directory *** :::::::::::::: pcm :::::::::::::: 00-00: USB Audio : USB Audio : playback 1 : capture 1 01-01: Bt87x Analog : Bt87x Analog : capture 1 01-00: Bt87x Digital : Bt87x Digital : capture 1 02-01: Bt87x Analog : Bt87x Analog : capture 1 02-00: Bt87x Digital : Bt87x Digital : capture 1 03-01: Bt87x Analog : Bt87x Analog : capture 1 03-00: Bt87x Digital : Bt87x Digital : capture 1 04-01: Bt87x Analog : Bt87x Analog : capture 1 04-00: Bt87x Digital : Bt87x Digital : capture 1 *** seq: directory *** :::::::::::::: timers :::::::::::::: G0: system timer : 1000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P1-0-1: PCM capture 1-0-1 : SLAVE P1-1-1: PCM capture 1-1-1 : SLAVE P2-0-1: PCM capture 2-0-1 : SLAVE P2-1-1: PCM capture 2-1-1 : SLAVE P3-0-1: PCM capture 3-0-1 : SLAVE P3-1-1: PCM capture 3-1-1 : SLAVE P4-0-1: PCM capture 4-0-1 : SLAVE P4-1-1: PCM capture 4-1-1 : SLAVE :::::::::::::: version :::::::::::::: Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:04.1[A] -> GSI 16 (level, low) -> IRQ 16 unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:05.1[A] -> GSI 17 (level, low) -> IRQ 17 unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:06.1[A] -> GSI 18 (level, low) -> IRQ 18 unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:07.1[A] -> GSI 19 (level, low) -> IRQ 19 ><> Nathan Stratton CTO, Voila IP Communications nathan at robotics.net nathan at voilaip.com http://www.robotics.net http://www.voilaip.com

2 3

[alsa-devel] pcm->private_data is NULL in alsa core!!
by Pharaoh . 22 Jun '07

22 Jun '07

Hi I am writing an alsa driver for my omap based board. I am testing it using cat /dev/audio currently, i.e. I have OSS emulation turned on. This is an initial round of testing and I will use alsa-libs and utils later. My problem is: I have set the pcm->private_data = my_chip after registering the pcm ops, after referring various standard drivers in alsa tree. My module is insmoded properly but after I do snd_card_register(card) the pcm->private_data is corrupted !! Before this step, the pcm->private_data is intact. But after looking at the snd_card_register(), I realized that it doesn't even touch the pcm pointer. So, when I cat the device I get huge segfault in snd_pcm_attach_substream(), since pcm->private_data is NULL but surprisingly all the other fields are intact in pcm pointer. Even the OSS code doesn't touch the pcm pointer, what must be wrong? I have checked my source atleast 5 times and couldn't find anything in it. -pharaoh.

2 14

[alsa-devel] AT91 SoC PCM Capture
by Paul Kavan 22 Jun '07

22 Jun '07

I am wondering what piece of code contains the capture functionality for the at91 soc. I am hoping to see how and when the pcm activates the ssc at read time. I just cannot figure out why TD0 would behave at such with this set of code and yet behave correctly with my polling code. Not a criticism....just wanting to figure this out. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 1

[alsa-devel] Osprey 440 almost supported!!! Need X9221 i2c support!
by Nathan Allen Stratton 22 Jun '07

22 Jun '07

This card does not use analog bt878a inputs, but rather the digital input. To get audio working I set the mask to 000303, first two bits are for the audio rate, 00 is 32000, 01 is 41000, 10 is 4800. The first two bits in the next byte are for balanced or unbalanced audio input. 0x00 for unbalanced and 0x100 for balanced. So I am using: modprobe bttv card=140 bttv_gpio=1 bttv_verbose=2 bttv_debug=1 gpiomask=0x020303 audioall=0xFFFDFC bttv: Bt8xx card found (0). bttv0: Bt878 (rev 17) at 0000:08:04.0, irq: 24, latency: 132, mmio: 0xc8600000 bttv0: detected: Osprey-440 [card=140], PCI subsystem ID is 0070:ff07 bttv0: using: Osprey 440 [card=140,insmod option] bttv0: gpio config override: mask=0x20303, mux=0xfffdfc,0xfffdfc,0xfffdfc,0xfffdfc bttv0: gpio: en=00000000, out=00000000 in=00feffff [pre-init] bttv0: risc main @ 37f25000 bttv0: gpio: en=00000000, out=00000000 in=00feffff [init] bttv0: BT878A ARESET bttv0: tveeprom i2c attach [addr=0x50,client=tveeprom] bttv0: using tuner=-1 bttv0: i2c: checking for TDA9887 @ 0x86... not found bttv0: registered device video0 bttv0: registered device vbi0 bttv0: gpio: en=00020303, out=00000000 in=00fcfcfc [audio: mute] bttv0: video mux: input=0 mux=2 bttv0: gpio: en=00020303, out=00000000 in=00fcfcfc [audio: mute] bttv0: PLL: 28636363 => 35468950 .<7>bttv0: gpio: en=00020303, out=00000000 in=00fcfcfc [audio: mute] ok So the issue now is that the audio is WAY to high so it is very distorted. The audio is controlled via X9221A chips on the i2c bus at 0x50, there are not drivers for X9221 so I can't lower the audio levels. Anyone able to help me with this last part??? ><> Nathan Stratton CTO, Voila IP Communications nathan at robotics.net nathan at voilaip.com http://www.robotics.net http://www.voilaip.com

2 2

[alsa-devel] Driver for Creamware / SonicCore Pulsar, which information needed?
by Kaspar Schleiser 22 Jun '07

22 Jun '07

Hey, I've got a Creamware Pulsar DSP card. As it actually has NO support for linux, I asked the vendor whether support was planned or if I could get necessary technical information. they answered they'd not have concrete plans on supporting linux and what information I'd need for the development of a driver. As I never wrote a driver, I don't know that, so maybe you could help me giving a useful answer ... What do we need to write a driver? Would anyone be interested in writing a driver? I could lend my card. It's a high quality multichannel card with 3 DSP chips on it. Cheers Kaspar

1 0

[alsa-devel] Steps get aplay and arecord working on an embedded board
by Pharaoh . 22 Jun '07

22 Jun '07

Hi I have to test my audio driver on an omap based board. I have got the lib and utils on my board but dont know the exact steps to follow for using aplay etc. Can anyone point me to relevant documentation? Couldn't find any. Or just brief outline about the steops to follow would be great help. -pharaoh

2 2

[alsa-devel] What is CONFIG_I2C_MODULE?
by Timur Tabi 21 Jun '07

21 Jun '07

In a lot of ALSA drivers, I see this: #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) What is CONFIG_I2C_MODULE? I've searched all the Kconfigs, and I cannot see this configuration option. -- Timur Tabi Linux Kernel Developer @ Freescale

1 0

[alsa-devel] HDA - Change default configuration for Fujitsu Si3515.
by Tobin Davis 21 Jun '07

21 Jun '07

This patch changes the default configuration for the Fujitsu Siemens Si3515 Laptop. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

1 0

[alsa-devel] Help with SoC alsa.conf
by Paul Kavan 21 Jun '07

21 Jun '07

I am trying to determine an appropriate conf file for my AT91SAM9260ek running linux-2.6.22-rc1 and the following alsa drivers: <6>Advanced Linux Sound Architecture Driver Version 1.0.14rc4 (Wed May 09 09:51:39 2007 UTC). <6>ASoC version 0.13.0 <4>AT91SAM9260EK W6811 Driver <1> <6>w6811: W6811 PCM SoC Audio 0.1 <6>asoc: W6811 <-> at91-ssc0 mapping ok <6>grh_ek_w6811: W6811 PCM in Slave Mode <6>ALSA device list: <6> #0: GRH_EK_W6811 (W6811) I have installed alsa-lib-1.0.14a and alsa-utils-1.0.14 on the at91sam9260. The default conf file produces sound using speaker-test and aplay, but will not record using arecord. Frank Mandarino has been kind enough to lend me a barebones conf that worked for him: pcm.!default { type hw card 0 } pcm.hw { type hw card 0 device 0 subdevice 0 } ctl.!default { type hw card 0 } ctl.hw { type hw card 0 } but produces the following for me: /usr/share/alsa $ aplay -l **** List of PLAYBACK Hardware Devices **** ALSA lib conf.c:3949:(snd_config_expand) Unknown parameters 0 ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0 aplay: device_list:213: control open (0): Invalid argument ALSA is *very* new to me and I am unsure how to interpret any of this. I am using a dumb W6811 pcm codec. I am fairly confident the machine/codec driver is working as expected, so I need to determine what I can do to get alsa to play nice with it as well. When I use the default alsa.conf produced from making alsa-libs, I get the following: /usr/share/alsa $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: GRHEKW6811 [GRH_EK_W6811], device 0: W6811 W6811-PCM-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 record and playing do not seem to work. When I try: /usr/share/alsa $ arecord -d5 test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono Aborted by signal Interrupt... It does not stop at 5 seconds....I have to interrupt it manually. Furthermore, it does not appear to record anything as when I: /usr/share/alsa $ aplay test.wav Playing WAVE ' test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono The speaker pops once, but no audio. Finally, when I use speaker-test, I get some weird results. Doing: /usr/share/alsa $ speaker-test -r8000 -c1 -f1000 -tsine -l1 speaker-test 1.0.14 Playback device is default Stream parameters are 8000Hz, S16_LE, 1 channels Sine wave rate is 1000.0000Hz Rate set to 8000Hz (requested 8000Hz) Buffer size range from 64 to 32768 Period size range from 32 to 8192 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Time per period = 5.086305 Gives me a wave composed of both the 1k sine and likely a 2k sine. If I go much below 1k, I get a periodic wave, but it looks more like a heart beat. My codec expects u-law, so that may be the problem with speaker-test. However, the fact the aplay and arecord are not working give me pause. I would be greatly appreciative of any insights, pointers, etc. If there is more information that would be helpful, please ask. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

1 3

Re: [alsa-devel] OSS/4Front
by Ash Willis 21 Jun '07

21 Jun '07

Hi Clemens, > You could compare <http://manuals.opensound.com/devlists/Linux.html> > with ALSA's sound card matrix, but the latter tends to be out of date. > Yeah, that thing is oooooold. Hopefully the new wiki on't be too bad. > > With the recent open sourcing of OSS, it would be nice to port code > > to ALSA for devices that are only currently covered by OSS. > > AFAICS all devices supported by the open-source OSS are also supported > by ALSA. > Ah, looks can be deceiving. Although both systems have 'geode' support, it seems that the OSS driver supports the CS5530 VSA2 and the SC1200. Also, there's an open bug for the Neomagic 256 driver, which may have a solution hiding away in OSS somewhere (someone with the hardware is hopefully going to help me look into that). There's a couple of things right off the bat and I don't know ALSA very well at all compared to most of you. And yeah, I know this stuff isn't in great demand, but it'd be nice if people weren't able to say that OSS can do XYZ and ALSA can't. Ash -- Get a Free E-mail Account at Mail.com! Choose From 100+ Personalized Domains Visit http://www.mail.com today

1 0

[alsa-devel] OSS/4Front
by Ash Willis 21 Jun '07

21 Jun '07

Hi, Does anyone know if there's any kind of list showing a comparison of hardware support between ALSA and OSS/4Front? With the recent open sourcing of OSS, it would be nice to port code to ALSA for devices that are only currently covered by OSS. There has to be at least a few devices that we can grab support for, right? It may at least provide us with insight into a few bugfixes. Ash -- Get a Free E-mail Account at Mail.com! Choose From 100+ Personalized Domains Visit http://www.mail.com today

4 3

[alsa-devel] Yamaha Cards
by derek＠darlingsong.net 21 Jun '07

21 Jun '07

Hi all, first post for me. I have a Yamaha DS2416 mixer card and Yamaha SW1000XG sound card that I would like to use onLinux (Fedora 6). I have used them on the "other" OS, along with ProTools, etc. Does anyone know if Linux drivers are available? I have looked around, and it seems not to be the case. any help would bve appreciated. Derekd Vancouver, BC

2 1

[alsa-devel] No time for cs4624
by chris＠idlelion.net 21 Jun '07

21 Jun '07

I really wanted to work on the cs4624 suspend resume bug (#305), but there is no time, especially given the lack of technical information. I'm wondering if I were to lend or give a laptop that exhibits the problem to one of you alsa developers, if that would be useful and provide some impetus to fix the bug, or if it would just be a big pain? Opinions? Chris

1 0

[alsa-devel] [1/2] 2.6.22-rc5: known regressions
by Michal Piotrowski 21 Jun '07

21 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc5. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : Kernel hang on CMOS_READ References : http://lkml.org/lkml/2007/6/3/138 Submitter : Rodrigo Luiz <troops.of.doom(a)gmail.com> Status : Unknown Subject : Device hang when offlining a CPU due to IRQ misrouting References : http://lkml.org/lkml/2007/5/31/419 Submitter : Darrick J. Wong <djwong(a)us.ibm.com> Handled-By : Siddha, Suresh B <suresh.b.siddha(a)intel.com> Status : problem is being debugged ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown FS Subject : 2.6.22-rc4-git5 reiserfs: null ptr deref. References : http://lkml.org/lkml/2007/6/13/322 Submitter : Randy Dunlap <randy.dunlap(a)oracle.com> Handled-By : Vladimir V. Saveliev <vs(a)namesys.com> Status : problem is being debugged PCI Subject : PCI setup hangs on Asus Notebook (nolapic helps) References : http://lkml.org/lkml/2007/6/14/161 Submitter : Malte Cornils <malte(a)cornils.net> Status : unknown Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

3 5

Re: [alsa-devel] Using aplay and arecord on an ARM
by Paul Kavan 20 Jun '07

20 Jun '07

On 6/20/07, Paul Kavan <pkavan(a)gmail.com> wrote: > > Okay: > > Here is where I ran into troubles yesterday. I am working on configuring > alsa-utils. When I do: > > CC=arm-linux-gcc ./configure --target=arm-linux --with-versioned=no > --prefix=/usr --host=i686-pc-linux-gnu > > I get the error: > > grhuser@pcL7:$ CC=arm-linux-gcc ./configure --target=arm-linux > --with-versioned=no --prefix=/usr --host=i686-pc-linux-gnu > configure: WARNING: If you wanted to set the --build type, don't use > --host. > If a cross compiler is detected then cross compile mode will be used. > checking for a BSD-compatible install... /usr/bin/install -c > checking whether build environment is sane... yes > checking for gawk... no > checking for mawk... mawk > checking whether make sets $(MAKE)... yes > checking for i686-pc-linux-gnu-strip... no > checking for strip... strip > checking whether NLS is requested... yes > checking for msgfmt... /usr/bin/msgfmt > checking for gmsgfmt... /usr/bin/msgfmt > checking for xgettext... /usr/bin/xgettext > checking for msgmerge... /usr/bin/msgmerge > checking for style of include used by make... GNU > checking for i686-pc-linux-gnu-gcc... arm-linux-gcc > checking for C compiler default output file name... a.out > checking whether the C compiler works... yes > checking whether we are cross compiling... yes > checking for suffix of executables... > checking for suffix of object files... o > checking whether we are using the GNU C compiler... yes > checking whether arm-linux-gcc accepts -g... yes > checking for arm-linux-gcc option to accept ISO C89... none needed > checking dependency style of arm-linux-gcc... gcc3 > checking build system type... i686-pc-linux-gnu > checking host system type... i686-pc-linux-gnu > checking for ld used by GCC... > /home/grhuser/Linux_Builds/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld > checking if the linker > (/home/grhuser/Linux_Builds/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld) > is GNU ld... yes > checking for shared library run path origin... done > checking for CFPreferencesCopyAppValue... no > checking for CFLocaleCopyCurrent... no > checking for GNU gettext in libc... no > checking for iconv... no, consider installing GNU libiconv > checking for GNU gettext in libintl... no > checking whether to use NLS... no > checking for i686-pc-linux-gnu-gcc... (cached) arm-linux-gcc > checking whether we are using the GNU C compiler... (cached) yes > checking whether arm-linux-gcc accepts -g... (cached) yes > checking for arm-linux-gcc option to accept ISO C89... (cached) none > needed > checking dependency style of arm-linux-gcc... (cached) gcc3 > checking for a BSD-compatible install... /usr/bin/install -c > checking whether ln -s works... yes > checking for ALSA CFLAGS... > checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread > checking for libasound headers version >= 1.0.12... not present. > configure: error: Sufficiently new version of libasound not found. > > I know there has to be a flag or something here I need to set. Does anyone > know what that might be? > > Paul > I think what occurs here is that the configure script is using my cross-compilers and associated libs in $HOME/Linux_Builds/buildroot/build_arm/staging_dir/ while the libs are actually given in $HOME/Linux_Builds/buildroot/build_arm/root/usr/lib/ I am looking at the README and INSTALL and do not see a solution to this. Does anyone have a solution to this? Thanks. Paul

4 11

[alsa-devel] Kernel 2.6.21 M-Audio Delta 1010L
by Suso Comesaña 20 Jun '07

20 Jun '07

Hello... My name is Suso Comesaña...I have 2 computers with a M-Audio card Delta 1010L...the problem is the following one: With kernel 2.6.17-2-multimedia-486... I can record sounds perfectly... but, kernel 2.6.21... I cannot record nothing... You have some idea?... You need that she sends some information to you? Tanks... Suso Comesaña. Peatón. susobaco(a)yahoo.es Usuario de Musix, GNU-Linux, basada en Debian/Knoppix. 100% Software Libre. ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com ____________________________________________________________________________________ ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. http://es.answers.yahoo.com/info/welcome

2 1

[alsa-devel] Output levels on SBLive! 5.1 on ALSA vs WIndows
by ChernYue Kwok 20 Jun '07

20 Jun '07

Hello, I am using an SBLive! 5.1 sound card (emu10k1 chipset) with 6 channel output. Due to the nature of my work, I need to be able to play sound through all 6 channels at the same output level to an amplifier. Using Linux and the emu10k1 ALSA driver, I measured output voltage levels on an oscilloscope while playing a sine wave on all 6 channels, and I got three different output voltage amplitudes for the front, rear and center/subwoofer channels. Repeating the same test with the same card on WIndows using the Creative SBLive drivers from their website, I am able to get the same output voltage amplitude on all 6 channels. For both tests, all output levels in the relevant mixers were set to their maximum levels. I would like to know if this output normalizing 'feature' of the SBLive Windows driver can be found in the emu10k1 ALSA driver, or in any other ALSA driver. Failing that, is there any other solution to set output levels in the driver, or am I doomed to twiddle in alsamixer to get the output levels that I want? Thanks in advance.

2 3

[alsa-devel] Using aplay and arecord on an ARM
by Paul Kavan 20 Jun '07

20 Jun '07

Liam: I have managed to get the alsa-libs and aplay, arecord, and speaker-test on my board, but I am unsure how to get them to play. I have tried making the device nodes and am using a simple alsa.conf, but still no joy. I am sure there is something simple I am missing. Trying to find some documentation on this. I am assuming there is some disconnect between the interface and my conf file. I have my alsa.conf file in /usr/share/alsa/ and it is: pcm.!default { type hw card 0 } pcm.hw { type hw card 0 device 0 subdevice 0 } ctl.!default { type hw card 0 } ctl.hw { type hw card 0 } ctl.w6811 { type hw card 0 } pcm.w6811 { type hw card 0 device 0 subdevice 0 } ctl.!default = ctl.w6811 pcm.!default = pcm.w6811 ctl.hw { @args[ CARD ] @args.CARD { type string default "0" } type hw card $CARD } pcm.hw { @args[ CARD DEV SUBDEV ] @args.CARD { type string default "0" } @args.DEV { type integer default 0 } @args.SUBDEV { type integer default 0 } type hw card $CARD device $DEV subdevice $SUBDEV } I have some /proc/ entries: /proc/asound $ ls GRHEKW6811 cards pcm version card0 devices timers /proc/asound $ cat cards 0 [GRHEKW6811 ]: W6811 - GRH_EK_W6811 GRH_EK_W6811 (W6811) /proc/asound $ cat devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer /proc/asound $ cat pcm 00-00: W6811 W6811-PCM-0 : : playback 1 : capture 1 /proc/asound $ cat timers G0: system timer : 10000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE /proc/asound $ cat version Advanced Linux Sound Architecture Driver Version 1.0.14rc4 (Wed May 09 09:51:39. Now, into GRHEKW6811 (which takes me to card0) I get a pcm0c and pcm0p directory. Using cat info, I get: /proc/asound/card0/pcm0c $ cat info card: 0 device: 0 subdevice: 0 stream: CAPTURE id: W6811 W6811-PCM-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 /proc/asound/card0/pcm0c $ cd ../pcm0p/ /proc/asound/card0/pcm0p $ cat info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: W6811 W6811-PCM-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 I then did: mknod -m 666 /dev/snd/pcm0p c 116 0 mknod -m 666 /dev/snd/pcm0c c 116 0 However, when I try speaker-test, I get: /usr/lib/bin $ ./speaker-test -D pcm0p speaker-test 1.0.14 Playback device is pcm0p Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM pcm0p Playback open error: -2,No such file or directory Same for default: /usr/lib/bin $ ./speaker-test -D default speaker-test 1.0.14 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory When I try aplay -l and arecord -l, I get: /usr/lib/bin $ ./aplay -l aplay: device_list:204: no soundcards found... /usr/lib/bin $ ./arecord -l arecord: device_list:204: no soundcards found... I am at a loss here. Anyone have any advice, pointers, tips, etc...... Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] ALSA always writes _whole_ periods, right?
by Joachim Förster 20 Jun '07

20 Jun '07

Hi ALSA devs, I have a quick question regarding the filling of the usual ring buffer in RAM (used by sound controllers via DMA), just to be really sure: Is it true, that ALSA always writes (via mmap'ing) _whole_ periods? I took snd-dummy and modified it to use "ack()" method and tried to confirm, if the above is true _without_ memory mapping (classical read/write). It seem's to be true, ALSA is "filling up" the last (not always complete) period - e.g. if the played wav file is not a multiple of periodsize in length. But well, with mmap, I cannot see the copying action (done by ALSA library, right?) ... so, thus my question ;-) . Thanks, Joachim

2 3

[alsa-devel] Surround problems with cs46xx
by Vasilis Manwlopoulos 20 Jun '07

20 Jun '07

Hi, Sorry for bothering the list again, but I have major problems with the surround 5.1 sound. I have manually compiled/installed th latest alsa-1.0.14 drivers but nothing improved. I am using Ubuntu 7.04 and my soundcard is the Hercules Fortissimo III. This is, as far as I know by reading on Internet, supported by the module snd-cs46xx in ALSA. I have enabled the surround sound by creating the following .asoundrc file. pcm.!default { type plug slave.pcm "surround51" slave.channels 6 route_policy duplicate } The problem is that sound pauses whenever I try to do something "heavy" such as opening a new firefox window or generally another app. Also, there are delays and echo between the front and the rear speakers from time to time. I think that the problem is that, this alsa setup(meaning surround5.1 and the specific soundcard) places a heavy load on the cpu. There are 99% cpu usage times when the only thing active is a player playing a "normal bitrate" mp3. When I delete the .asoundrc file and I switch to default stereo sound using only the 2 front speakers the problems disappear. Can anyone suggest anything about this? Thanks in Advance V.Manolopoulos

3 3

[alsa-devel] Kernel 2.6.21 M-Audio Delta 1010L
by Suso Comesaña 20 Jun '07

20 Jun '07

Hello... My name is Suso Comesaña...I have 2 computers with a M-Audio card Delta 1010L...the problem is the following one: With kernel 2.6.17-2-multimedia-486... I can record sounds perfectly... but, kernel 2.6.21... I cannot record nothing... You have some idea?... You need that she sends some information to you? Tanks... Suso Comesaña. Peatón. susobaco(a)yahoo.es Usuario de Musix, GNU-Linux, basada en Debian/Knoppix. 100% Software Libre. ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com ____________________________________________________________________________________ ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. http://es.answers.yahoo.com/info/welcome

1 0

[alsa-devel] [PATCH] Output MFG information for HDA devices
by Jonathan Phénix 19 Jun '07

19 Jun '07

Hi, this patch output information about the MFG (Modem Function Group) to the /proc/asound/card?/codec#? file. This information is important to identify if a HDA device has a modem or not. Regards, Jonathan Phénix Technical specialist / Linuxant www.linuxant.com jonathan(a)linuxant.com

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Re: [alsa-devel] Writing an AT91SAM9260-EK Dumb Codec Driver
by Paul Kavan 19 Jun '07

19 Jun '07

On 6/19/07, Liam Girdwood <lg(a)opensource.wolfsonmicro.com> wrote: > > On Mon, 2007-06-18 at 14:30 -0500, Paul Kavan wrote: > > > > > Am I at a point now where I can try to get alsa-lib and alsa-util in > > place? If so, what version? I found some tars at: > > > > http://www.alsa-project.org/alsa/ftp/lib/ > > > > Is that the best place to get them? I am using buildroot, but not sure > > if ALSA stuff is ready for primetime there. I guess I should check on > > this asap. > > > > Afaik, there is an issue with buildroot atm (please someone correct me > if it's now fixed) when it comes to building alsa-lib for ARM. (please > search archives for details). No, I think you are right. I have tried a few recommended patches, but they do not work. Fwiw, I use openembedded for all my embedded builds without any problems > atm. http://www.openembedded.org At some point when I have some time, I am going to give it a try. I have managed to build the libs and have those in my rfs using: CC=arm-linux-gcc ./configure --host=arm-linux --prefix=$HOME/buildroot/build_arm/root/usr --with-versioned=no ...and then... make install With this, I get the /usr/ entries in my rfs. However, I have had troubles building alsa-utils. When I try: CC=arm-linux-gcc ./configure --host=arm-linux --prefix=$HOME/buildroot/build_arm/root/usr --with-versioned=no I get: grhuser@pcL7:$ CC=arm-linux-gcc ./configure --host=arm-linux --prefix=$HOME/buildroot/build_arm/root/usr --with-versioned=no configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... no checking for mawk... mawk checking whether make sets $(MAKE)... yes checking for arm-linux-strip... arm-linux-strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for arm-linux-gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gcc accepts -g... yes checking for arm-linux-gcc option to accept ISO C89... none needed checking dependency style of arm-linux-gcc... gcc3 checking build system type... i686-pc-linux-gnu checking host system type... arm-unknown-linux-gnu checking for ld used by GCC... /home/grhuser/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld checking if the linker (/home/grhuser/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking for CFPreferencesCopyAppValue... no checking for CFLocaleCopyCurrent... no checking for GNU gettext in libc... no checking for iconv... no, consider installing GNU libiconv checking for GNU gettext in libintl... no checking whether to use NLS... no checking for arm-linux-gcc... (cached) arm-linux-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gcc accepts -g... (cached) yes checking for arm-linux-gcc option to accept ISO C89... (cached) none needed checking dependency style of arm-linux-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.12... not present. configure: error: Sufficiently new version of libasound not found. Not sure what is the problem here. I have file:///home/grhuser/buildroot/build_arm/root/usr/lib/libasound.so.2.0.0 in my rfs. Any suggestions? Paul Liam > > -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 1

[alsa-devel] [PATCH] hda: add eapd support to additional idt codecs
by Matthew Ranostay 19 Jun '07

19 Jun '07

Added support for EAPD on the 927x, and 9227-8 IDT HDA codecs. Enabling EAPD powers the internal speaker amp otherwise there is no sound on systems with an internal amp. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 96495fe7cc13 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jun 14 12:58:23 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Mon Jun 18 17:10:38 2007 -0400 @@ -832,6 +832,18 @@ static void stac92xx_set_config_regs(str } } +static void stac92xx_enable_eapd(struct hda_codec *codec) +{ + /* Configure GPIO0 as output */ + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DIRECTION, 0x00000001); + /* Configure GPIO0 as CMOS */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); + /* Assert GPIO0 high */ + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, 0x00000001); + /* Enable GPIO0 */ + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_MASK, 0x00000001); +} + /* * Analog playback callbacks */ @@ -2193,7 +2205,8 @@ static int patch_stac927x(struct hda_cod } spec->multiout.dac_nids = spec->dac_nids; - + stac92xx_enable_eapd(codec); + err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) { @@ -2254,18 +2267,7 @@ static int patch_stac9205(struct hda_cod spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - - /* Configure GPIO0 as EAPD output */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + stac92xx_enable_eapd(codec); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) {

2 1

[alsa-devel] Is snd_magic_kcalloc() deprecated?
by Pharaoh . 19 Jun '07

19 Jun '07

2 1

[alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Matt Mullins 19 Jun '07

19 Jun '07

I posted this to the LKML, I thought it should be reproduced on this list. > > On Sunday, 17 June 2007 03:36, Matt Mullins wrote: > > > I just received a Dell Latitude D630, with the new Intel Santa Rosa > > > platform. Currently, the only major driver issue I have is sound. It > > > worked fine in Ubuntu Feisty's 2.6.20 kernel, but now I am using Gutsy > > > so I can have graphics drivers. Gutsy's 2.6.22-rc3-based kernel no > > > longer recognized my soundcard, a Sigmatel STAC9205, which uses the > > > hda_intel driver. I have since fixed that problem by compiling the > > > hda_intel driver into the kernel (I used 2.6.22-rc4 vanilla sources), > > > instead of a module, but it is still a bug that it does not work as a > > > module. > > > > > > All the following debug information was obtained from kernel 2.6.22-rc4. > > > > > > However, the bug that is currently affecting me is that sound no > > > longer works after using ACPI suspend-to-RAM or swsusp. I compiled > > > ALSA debugging in, and as soon as I resume, I get tons of messages > > > like: > > > [ 8.474830] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0970500 > > > [ 8.474954] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0af0009 > > > [ 8.475078] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0a70500 > > > [ 8.475207] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0bf0009 > > > (etc) > > > > > > There may be more before that, but that is more than the kernel > > > message buffer can hold, so I can't ever see it. I get similar > > > timeout messages each time a PCM gets set up: > > > [ 8.972859] hda_codec_setup_stream: NID=0x10, stream=0x5, channel=0, format=0 > > > x11 > > > [ 8.972987] hda-intel: send_cmd timeout: IRS=0x1, val=0x1070650 > > > [ 9.006071] hda-intel: send_cmd timeout: IRS=0x1, val=0x1020011 > > > [ 9.006081] hda_codec_setup_stream: NID=0x11, stream=0x5, channel=0, format=0 > > > x11 > > > [ 9.006207] hda-intel: send_cmd timeout: IRS=0x1, val=0x1170650 > > > [ 9.016117] hda-intel: send_cmd timeout: IRS=0x1, val=0x1120011 > > > > > > I read some of sound/pci/hda/hda_intel.c, specifically the code that > > > output those messages. After octupling my kernel log buffer (to 1MB), > > > I noticed that for the first set of timeouts (immediately upon > > > resume), all the "codec" nibbles are odd, the last word of the val > > > ("verb" and "para") are all either 0x0500 or 0x0009. For the second > > > set (when something tries to use the card) that last word is 0x0650 or > > > 0x0011. > > > > > > I have yet to go back and re-test compiling snd-hda-intel as a module. On 6/17/07, Matt Mullins <mokomull(a)gmail.com> wrote: > I've been reading up on the Intel HD Audio specifications, and I found > that the problem is most likely that the driver has to resort to PI/O > (single_cmd) mode. I really haven't had enough time or experience to > figure out what is actually causing this. This is the first time I've > done any kernel development; thankfully, Intel has good documentation. > I'm going to keep hacking to see what information I can figure out. > > I just tried 2.6.22-rc5, and no behavior has changed.

3 3

[alsa-devel] Bad digital out of ASUS M2V
by Mike Crash 19 Jun '07

19 Jun '07

In ALSA 1.0.14 HDA Intel with ASUS M2V motherboard - detected the chip ALC660VD, but digital out is missing. I added workaround - added new model name 3stack-660-digout with digital output, diff file follows: ---------------------------------------------- 117a118 > ALC660VD_3ST_DIG, 9466a9468 > [ALC660VD_3ST_DIG]= "3stack-660-digout", 9478c9480 < SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), --- > SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), 9501a9504,9516 > [ALC660VD_3ST_DIG] = { > .mixers = { alc861vd_3st_mixer }, > .init_verbs = { alc861vd_volume_init_verbs, > alc861vd_3stack_init_verbs }, > .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), > .dac_nids = alc660vd_dac_nids, > .dig_out_nid = ALC861VD_DIGOUT_NID, > .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), > .adc_nids = alc861vd_adc_nids, > .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), > .channel_mode = alc861vd_3stack_2ch_modes, > .input_mux = &alc861vd_capture_source, > },

2 1

[alsa-devel] question about bluez using alsa and patches
by Walt Shekrota 19 Jun '07

19 Jun '07

I was told that when using the btsco(bluez) and/or plugs bluetooth headset software a patch is always required. In the Gentoo community and perhaps everywhere it is suggested to use alsa compiled into your kernel. As I go to new kernels frequently these folks (sco headset) seem to have no such forward support ie they have not worked on the patches yet for the newer kernels. Am I wrong or at some point in time should this app not be weaned from using patches? Is it the fact they use sco that they have to patch? Is there not an alsa library they can program to instead. I guess I don't understand the overall design so I can't see why this goes on forever. I'm somewhat apprehensive about trusting others to continually modify my kernel. Am I out in left field here? Can anyone explain why they must choose this path? Thanks very much. -Walt

3 3

[alsa-devel] Difference between substream->dma_buffer and substream->runtime->dma_buffer_p?
by Timur Tabi 19 Jun '07

19 Jun '07

The snd_pcm_substream structure has a snd_dma_buffer structure. snd_dma_buffer looks like this: struct snd_dma_buffer { struct snd_dma_device dev; /* device type */ unsigned char *area; /* virtual pointer */ dma_addr_t addr; /* physical address */ size_t bytes; /* buffer size in bytes */ void *private_data; /* private for allocator; don't touch */ }; snd_pcm_substream also has a snd_pcm_runtime structure, which looks like this: struct snd_pcm_runtime { ... /* -- DMA -- */ unsigned char *dma_area; /* DMA area */ dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ size_t dma_bytes; /* size of DMA area */ struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ Why the redundancy? Not only does snd_pcm_runtime have dma_area, dma_addr, and dma_bytes, but it also has a pointer to a dns_dma_buffer structure, which also has this information! So given a snd_pcm_substream structure, you can obtain the physical address of the DMA buffer three ways: substream->dma_buffer.addr substream->runtime->dma_addr substream->runtime->dma_buffer_p->addr When will these three variable contain different values? If the answer is never, then why do they all exist? -- Timur Tabi Linux Kernel Developer @ Freescale

3 2

[alsa-devel] [PATCH] ASoC S3C24xx machine drivers - lm4857-h missing patch
by Graeme Gregory 19 Jun '07

19 Jun '07

Somewhere when dealing with the s3c24xx patches and requested changes I managed to miss this file on re-submission. Its a header file required for compilation of neo1973-wm8573.c and should be a no brainer. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Graeme Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

2 1

[alsa-devel] Changing the sample rate/channels of an already opened PCM stream
by Ed Schouten 19 Jun '07

19 Jun '07

Hello, I am the author of the command line music player Herrie (http://herrie.info/). For a long time folks have asked me to add support for OSS to Herrie, so this morning I decided to install Linux on a spare disk of mine (Ubuntu 7.04, FWIW) to hack on ALSA support for Herrie. After (a real lot of) hours of reading code, reading the Doxygen stuff, etc. I now have a functional ALSA output module for Herrie. It was quite hard to get it working, so I guess I owe myself a beer! ;-) Anyway, there still is one disadvantage to my module; each time I detect that a new track is going to be started that needs a different amount of channels or sample rate, I reopen the ALSA device, something that is just impossible. The reason why it's impossible: Herrie has a nifty chroot() functionality that allows users to trap the application in a certain directory. I have been trying to get the device to reset so I can change the sample rate, but for some reason, it just gets ignored. Can someone explain me how it's possible to reset the sample rate after preparing the handle? For your information, here's a link to the latest version of the source code of my ALSA module: http://g-rave.nl/gitweb?p=herrie;a=blob;f=herrie/src/audio_output_alsa.c Yours, -- Ed Schouten <ed(a)fxq.nl> WWW: http://g-rave.nl/

2 2

[alsa-devel] Writing an AT91SAM9260-EK Dumb Codec Driver
by Paul Kavan 18 Jun '07

18 Jun '07

Hello all: I am working on a driver for a dumb codec to work with the SSC as a PCM on the AT91SAM9260. I have, with the gracious help of Frank Mandarino, have the glue code written without too many problems. However, I need some guidance on getting a few things configured correctly for my codec and then getting it to compile and load correctly. For configuration, I am using the Winbond W6811. For this, I am going to use a 256kHz BCLK and an 8kHz frame. Each sample consists of 8-bit u-law PCM, MSB first. When I execute a read from the codec to the processor, I need to make the reads on the falling edge of the BCLK at each sync pulse. When executing a write from the processor to the codec I need to write on the rising edge at each sync pulse. I think I the timing correct in my driver with: cmr_div = 187; /* PCM_CLK = ~96MHz/(2*187) = 256kHz */ period = 15; /* PCM_SYNC = PCM_CLK/(2*(15+1)) = 8000Hz */ However, I am not sure how to get the following correct for my needs: #define W6811_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) Now to making the driver once the code is done. I am relatively new to writing linux drivers. In the 6 or so that I have written, I usually cross-compile them as modules and then insmod them. I have tried that here and am getting a strange warning that I am unsure of. I have a pretty simple Makefile and when I execute it, I get: make ARCH=arm CROSS_COMPILE=arm-linux- make -C ~/Linux_Builds/linux-2.6.21.5M=/home/grhuser/Door_Access_System/Remotes/Driver/sound modules make[1]: Entering directory `/home/grhuser/Linux_Builds/linux- 2.6.21.5' CC [M] /home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.o Building modules, stage 2. MODPOST 1 modules WARNING: "at91_ssc_dai" [/home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.ko] undefined! CC /home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.mod.o LD [M] /home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.ko make[1]: Leaving directory `/home/grhuser/Linux_Builds/linux- 2.6.21.5' As you can see, I am using linux-2.6.21.5 I have ALSA checked and AT91-SoC checked in my .config file. For this attempt to build the driver, I have a separate folder than contains: at91-ssc.h and at91-pcm.h. If there is a more correct way to do this and be able to debug it, please let me know. I have attached copies of my driver. I would appreciate any and all feedback and assistance on getting this running.

2 6

[alsa-devel] .asoundrc
by Nobin Mathew 18 Jun '07

18 Jun '07

Hi, I am trying to write one .asoundrc file What i am currently doing is rate conversion My .asounrc file contains pcm.sirf3a { type rate slave { pcm "hw:0,0" rate 8000 } } I am using following command aplay -M -D sirf3a /usr/share/sounds/alsa/Side_Right.wav But it is giving me some errors aplay: set_params:909: Channels count non available Any idea why this? where I am going wrong?

2 2

[alsa-devel] [ASoC] Missing shutdown() vectors in ASoC
by Nobin Mathew 18 Jun '07

18 Jun '07

Liam, Shutdown() vectors is missing in ASoC, we need this to ensure proper shutdown of the system. * codec device */ struct snd_soc_codec_device { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, pm_message_t state); int (*resume)(struct platform_device *pdev); }; /* SoC CPU DAI */ struct snd_soc_cpu_dai { /* DAI description */ char *name; unsigned int id; unsigned char type; /* DAI callbacks */ int (*probe)(struct platform_device *pdev); void (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); unsigned int (*config_sysclk)(struct snd_soc_cpu_dai *cpu_dai, struct snd_soc_clock_info *info, unsigned int clk); /* DAI capabilities */ struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; struct snd_soc_dai_cap caps; /* SoC platform interface */ struct snd_soc_platform { char *name; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); /* pcm creation and destruction */ /* SoC machine */ struct snd_soc_machine { char *name; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); /* the pre and post PM functions are used to do any PM work before and * after the codec and DAI's do any PM work. */ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); int (*suspend_post)(struct platform_device *pdev, pm_message_t state); int (*resume_pre)(struct platform_device *pdev); int (*resume_post)(struct platform_device *pdev); Nobin Mathew

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[alsa-devel] Query regarding pcm->private_data
by Pharaoh . 18 Jun '07

18 Jun '07

Hi I am reading 'Wring an alsa driver'. It says following about the private_data: You can allocate a record for the substream and store it in runtime->private_data. Usually, this is done in the open callback. Don't mix this with pcm->private_data. The pcm->private_data usually points the chip instance assigned statically at the creation of PCM, while the runtime->private_data points a dynamic data created at the PCM open callback. What does static and dynamic mean here? Apologies for this question if it sounds very novice, since my understanding is it doesn't mean static and dynamic in the usual way like stack and heap etc. Why I am being so paranoid about this has reason: In my ALSA driver I am assigning pcm->private_data = mychip, but it is NULL in the snd_pcm_attach_substream() function. I have checked my code many times and I tried with another driver which was working when tested using ALSA utils. Currently I am testing using the OSS emulation i.e. simply cat /dev/audio.

2 1

[alsa-devel] Alsa mixer interface limitations?
by Ctirad Fertr 17 Jun '07

17 Jun '07

Hello! I'm currently in phase of adding support for higher sample rates into e-mu 1212m/1820m driver. These cards have a lot of physical inputs and outputs and also a bunch of internal buses used for capture, playback and internal routing between physical inputs and outputs. Both physical I/O and internal buses are shown in alsamixer, so one can set routing easily. It's a lot of switches, anyway. The card has three modes. The standard (or "single speed") mode is the only mode currently supported in emu10k driver. It is capable of 44.1/48kHz sample rate only. If we need higher sample rates, we have to switch the card into either "double speed" (for 88.2/96kHz) or "quadruple speed" (for 176.4/192kHz) mode. The problem is that the number of inputs, outputs and especially internal buses significantly changes at higher speed modes. The reason is that internal buses are 44.1/48kHz only and thus we must use two (or four) of them per channel. Also some physical I/O ports disappear at double and quadruple speeds (e.g. ADAT has 4 channles at 96k and 2 at 192k, S/PDIF is completely disabled at 192k). The question is how to deal with that in alsamixer? IMHO the best way would be to change the number of switches in alsamixer on the fly. Unfortunatelly, I'm afraid it's impossible. Another way would be to make a particular swiches inactive, though I don't know if it is a good idea and it wold require a lot of code anyway. Any ideas? Please help! Ctirad

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[alsa-devel] Got microphone working with SigmaTel STAC 9221 A1 on a Dell XPS M1210
by Eran Guendelman 17 Jun '07

17 Jun '07

I installed SuSE 10.2 on my new XPS M1210 laptop and found that the microphone was not working (headphones and internal speakers worked fine). I eventually got it working, and wanted to write an email describing how I did this. The two main important steps (which I will describe in more detail below) were: - Changing the default pin configurations ref922x_pin_configs in patch_sigmatel.c to values that were based on the windows driver's INI file but with one important tweak. - Running the driver using the "model=ref" parameter so that the above modified pin configurations are used. Note that it could be that my solution is overkill and there's actually a way to get this working with the unmodified alsa drivers... But I did not find any such solutions (e.g. I've tried setting "model=ref", "model=3stack", "model=5stack", but none of those fixed the microphone). In general, it seems that the solutions proposed on the internet to getting the SigmaTel codec working are very dependent on the exact type of system you've got... So this will likely only work for a Dell XPS M1210. Here is some more system info: "lspci -v" gives this as my audio card: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) Subsystem: Dell Unknown device 01d7 Flags: bus master, fast devsel, latency 0, IRQ 233 Memory at efffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) My card uses the snd_hda_intel module, and the codec type (given by "cat /proc/asound/card0/codec#0") gives this information at the top: Codec: SigmaTel STAC9221 A1 Address: 0 Vendor Id: 0x83847680 Subsystem Id: 0x102801d7 Revision Id: 0x103601 So here's what I did (taking many days and many wrong steps before arriving at this solution): - I have a dual-boot system and installed the official DELL drivers for the sound card in Windows XP. My goal was to find which INI file corresponded to my installed sound card, so I did the following: - Using Windows device manager, I looked at the properties for the SigmaTel codec driver and found that one of the property items had a string that included the substring VEN_8384&DEV_7680&SUBSYS_102801D7 - The self-extracting DELL driver extracted files into c:\dell\drivers\r122161, and in the wdm subdirectory I searched for the above string and found that the files 92xxM.INI, Sthda.ini, and sthda64.ini all contained the above string and also had the suffix =92xxM8-2.ini appended to it. This led me to deduce that the relevant file was 92xxm8-2.ini - This is an excerpt from 92xxm8-2.ini: [HKR\Settings\pin\0A] CfgDflt = dword: 0x0221121E AltCfg = hex: 20,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\pin\0B] CfgDflt = dword: 0x408103FF [HKR\Settings\pin\0C] CfgDflt = dword: 0x0281123E AltCfg = hex: 22,12,11,02,30,12,A1,02 Bias = hex: FF,04,02 Action: hex: 0A, 0C [HKR\Settings\pin\0D] CfgDflt = dword: 0x90100310 [HKR\Settings\pin\0E] CfgDflt = dword: 0x408003F1 [HKR\Settings\pin\0F] CfgDflt = dword: 0x0221122F AltCfg = hex: 21,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\pin\10] CfgDflt = dword: 0x03451340 [HKR\Settings\pin\11] CfgDflt = dword: 0x40C003F2 [HKR\Settings\pin\15] CfgDflt = dword: 0x50A003F3 [HKR\Settings\pin\1B] CfgDflt = dword: 0x405003F4 - This will be useful later... Now back to linux. - I downloaded latest alsa-driver package (version 1.0.14). - I edited alsa-kernel/pci/hda/patch_sigmatel.c, and replaced the contents of the ref922x_pin_configs array with the default pin configurations found in the INI file above WITH ONE IMPORTANT TWEAK. Specifically, this is my new array: static unsigned int ref922x_pin_configs[10] = { 0x0221121E, 0x408103FF, 0x02A1123E, 0x90100310, 0x408003F1, 0x0221122F, 0x03451340, 0x40C003F2, 0x50A003F3, 0x405003F4 }; - If you compare these numbers to the CfgDflt values from the Windows INI file above, you'll see that they match except that for the third value, instead of 0x281123E I use 0x2A1123E Here's why: I figured out that pin complex 0x0c was my microphone jack (the right-most of the three jacks in the front of my laptop). The 0x281123E configuration indicates a "Line In at Ext Front" jack... Indeed, this jack can serve as a line in or mic in or even I believe as output to rear speakers. But designating it as Line In by default seems to confuse the patch_sigmatel.c routines that set up the codec... They assume it's just a line in jack and do not mark it as potential microphone input. Changing the 8 to an A in the value designates it as "Mic at Ext Front", so the codec recognizes it as the microphone input for the card. What ends up happening is that as a microphone input, the Pin-ctls for the pin gets assigned to 0x24, whereas when it just thinks it's a Line In the Pin-ctls gets set to 0x20. Both designate an input, but the 4 in 0x24 seems to also give it a Voltage Reference Signal which I guess is important for microphone inputs. (I don't fully understand most of this stuff... I've just been reading up about the HD Audio specs over the last few days). - Build and install drivers using "./configure --with-cards=hda-intel --with-sequencer=yes; make; make install" - Make sure the snd_hda_intel gets loaded with model=ref option. You need to specify the ref model so that the pin configurations set up above are used. Of course, the right way to do this would be to make a whole new model, but I didn't bother doing that in my quick hack. Anyway, in SuSE 10.2 you can set the model through YaST, or you can manually edit /etc/modprobe.d/sound and make sure model=ref is listed as one of the options in the "options snd-hda-intel . . ." line - Reset sound drivers. (Best to just reset computer, but on SuSE 10.2 it usually works if I do "/etc/init.d/alsasound reset") - And that's basically it... In the mixer (e.g. alsamixer) it's important to make sure the Capture channel has "CAPTUR" on and that it's at a high value, and that the Mux is at its max too. It does not allow me to change "Input Source", but that's fine because the above essentially hard codes it to be Front Mic which is what I wanted. I've attached the contents of /proc/asound/card0/codec#0 in its good (mic working) state. The final path is 0x0c (mic) -> 0x12 (audio selector, stereo amp-out; Mux in the mixer) -> 0x17 (audio selector, stereo amp-in; Capture in the mixer) -> 0x06 (audio input) After searching on the internet, reading about HD Audio, and poking around the driver code, I do have a number of lingering questions I would be happy to have answered: - Why does patch_sigmatel.c "fix" mux capture levels to 2? The Mux (0x12) ends up in the path from the mic in jack to the audio in, so is it supposed to act as a Mic Boost or something? - I noticed my headphone jacks (0x0a, 0x0f) have Pin-ctls set at 0xc0 which is OUT HP (i.e. uses a low impedance amplifier for headphone output). Would it damage my sound card or speakers if I connected these jacks to powered speakers or to an audio receiver (i.e. more traditionally connected to a Line Out)? - Does the snd_hda_intel driver currently (or will it eventually) support dynamic switching so I can switch from the Mic In to Line In or to Rear Speakers Out, etc, just like in the Windows driver? - Some solutions on the internet talked about creating an .asoundrc file to get the codec working... I don't understand what that file is for and whether I need to tweak it at all... But since I have a working solution without the file I guess I'm fine? Thanks, Eran. P.S. Not subscribed, so please respond by email... Codec: SigmaTel STAC9221 A1 Address: 0 Vendor Id: 0x83847680 Subsystem Id: 0x102801d7 Revision Id: 0x103601 Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0e, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x6b 0x6b] Power: 0x0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x66 0x6d] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x17 Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x18 Node 0x08 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x09 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x11 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Pin Default 0x0221121e: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x408103ff: [N/A] Line In at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a1123e: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Pin Default 0x90100310: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x408003f1: [N/A] Line In at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x0221122f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x05 Node 0x10 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x03451340: [Jack] SPDIF Out at Ext Left Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 3 0x08* 0x17 0x19 Node 0x11 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x40c003f2: [N/A] SPDIF In at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x02 0x02] Connection: 7 0x0e 0x15 0x0f 0x0b 0x0c* 0x0d 0x0a Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 7 0x0e 0x15 0x0f 0x0b 0x0c* 0x0d 0x0a Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x50a003f3: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Volume Knob Widget] wcaps 0x600000: Mono Node 0x17 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x0e 0x0e] Connection: 1 0x12 Node 0x18 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x80 0x80] Connection: 1 0x13 Node 0x19 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Node 0x1a [Audio Output] wcaps 0x30201: Stereo Digital Node 0x1b [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x405003f4: [N/A] Digital Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x1a

2 5

[alsa-devel] Sound problem in Toshiba
by Furkan Cayci 17 Jun '07

17 Jun '07

Hi, I am using Ubuntu 7.04. After installing Ubuntu, I could't hear any sound from my Toshiba Satellite A205-S4577, so I did the things related with this article; http://pastebin.ca/522731 After restarting laptop, I didn't take any errors, but there was no sound again. Then I realized the output of the dmesg. $ dmesg|grep alsa [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 19.672000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 19.680000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) [ 33.572000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:1810: hda_codec: model 'auto' is selected [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 34.360000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 34.368000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) ------------------------- So, there is something wrong, but I didn't understand and I was wondering if you could solve or show a way to do it. Also, here is the output of lcpci -v "00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Toshiba America Info Systems Unknown device ff10 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at f0b40000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0" Thanks for your help, Furkan

1 0

[alsa-devel] Software mixing according to priority of streams
by Nobin Mathew 15 Jun '07

15 Jun '07

Hi, Is there any sound mixer which does mixing according to priority of streams, i.e. i will have streams with different priority like audio stream , bluettoth sco stream, gps navigation sound stream etc. Since my hardware will not support hardware mixing i will have to do this mixing in software. If bluetooth sco stream is opened while audio stream is playing then the application (sound server) should stop the playback of audio stream and should start playback of bletooth sco stream. priority order is bluettoth sco > gps navigation sound > audio Can dmix and Jack server the purpose. Nobin Mathew

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[alsa-devel] [RFC][ASoC] New device/driver arch
by Liam Girdwood 15 Jun '07

15 Jun '07

I'd like to gather some feedback on some changes I've been making to ASoC recently. The changes were made to fit in better with the PPC device tree and to help with easier porting to new platforms. They were also introduced to make dealing with device probe errors (e.g. I2C probe failure) a little easier. The largest change is the introduction of an ASoC bus device. All component drivers (that is, codec, codec DAI, CPU DAI, and DMA PCM drivers) register with this bus. This should simplify device registration and hopefully better fit in with the PPC model. A component driver now typically consists of the standard driver ops and a set of ASoC/ALSA ops to configure audio. The new soc.h shows this here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Example component drivers are here :- I2S http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… DMA http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… WM8753 Codec http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Another major change is that there is no more snd_dai_link (structure that connected codec <--> CPU). This has been replaced by an ASoC PCM device that describes each PCM device in the machine. This new ASoC PCM device has it's own driver and registers with the bus (like the others). This allows an ASoC PCM driver to be defined like :- static struct snd_soc_device_driver hifi_pcm_drv = { .type = SND_SOC_BUS_TYPE_PCM, .driver = { .name = "mainstone-hifi-pcm", .owner = THIS_MODULE, .probe = hifi_pcm_probe, .remove = hifi_pcm_remove, }, .components = { .cpu_dai = pxa2xx_i2s, .codec = wm8753_codec, .codec_dai = wm8753_hifi_dai, .platform = pxa2xx_pcm, }, }; This now shows each component within the driver structure. The snd_soc_machine driver has changed and can now contain numerous ASoC PCM devices. It is now a platform driver. Example machine driver for Mainstone with WM8753 is here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… The machine driver now also does the I2C/SPI probing (instead of the codec) and also does any codec IO (to better support multiple controllers). Fwiw, I've not tested any of this code atm. I plan to start this tomorrow on pxa2xx and i.MX31 and would like to get some feedback before porting remaining platforms and codecs over. Liam

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[alsa-devel] Documentation for params_xxx() macros?
by Timur Tabi 14 Jun '07

14 Jun '07

Is there documentation that describes each of the params_xxx() macros? If not, then which macro tells me how many periods there are? Is it params_periods()? -- Timur Tabi Linux Kernel Developer @ Freescale

1 0

[alsa-devel] aplay hangs with execve("/build/bin/aplay", ["aplay"], [/* 8 vars */
by Pharaoh . 14 Jun '07

14 Jun '07

Hi I am using alsa lib and alsa utils 1.0.14 on my omap based board. When I try to test my my alsa driver using aplay it just hangs at the very first step. The strace output shows nothing else except the following line, execve("/build/bin/aplay", ["aplay"], [/* 8 vars */ What I am missing?

1 0

[alsa-devel] Drivers for M-Audio Firewire 410
by Cédric 14 Jun '07

14 Jun '07

Hello, Im a french user of Linux (Ubuntu Studio 7.04) and cant install my new hardware (M-Audio Firewire 410). Ive send a mail to the M-Audios technical support, but they have no solution. They recommended me to contact yourself. Could you help me for this problem. Do you need more informations about my computer hardware ? Syncerly. Cédric

2 1

[alsa-devel] [PATCH]is_power_of_2-sound/pcm_params.h
by vignesh babu 14 Jun '07

14 Jun '07

Replacing (n & (n-1)) in the context of power of 2 checks with is_power_of_2 Signed-off-by: vignesh babu <vignesh.babu(a)wipro.com> --- diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 85cf1cf..dee298e 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -21,6 +21,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ +#include <linux/log2.h> int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, @@ -170,7 +171,7 @@ static inline int snd_mask_single(const struct snd_mask *mask) for (i = 0; i < SNDRV_MASK_SIZE; i++) { if (! mask->bits[i]) continue; - if (mask->bits[i] & (mask->bits[i] - 1)) + if (!is_power_of_2(mask->bits[i])) return 0; if (c) return 0; -- Vignesh Babu BM _____________________________________________________________ "Why is it that every time I'm with you, makes me believe in magic?" The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[alsa-devel] [PATCH] Fix build error with strict mm checking
by Martin Habets 13 Jun '07

13 Jun '07

Turning on STRICT_MM_TYPECHECKS in include/asm/page.h causes a build error for sparc32. This patch against 2.6.21 fixes this. Martin Signed-off-by: Martin Habets <errandir_news(a)mph.eclipse.co.uk> --- --- linux/sound/core/sgbuf.c.orig 2007-06-06 23:53:34.000000000 +0100 +++ linux/sound/core/sgbuf.c 2007-06-06 23:54:45.000000000 +0100 @@ -96,7 +96,7 @@ void *snd_malloc_sgbuf_pages(struct devi } sgbuf->size = size; - dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); + dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, __pgprot(PAGE_KERNEL)); if (! dmab->area) goto _failed; return dmab->area;

2 1

[alsa-devel] Trouble understanding ALSA's DMA buffers
by Timur Tabi 12 Jun '07

12 Jun '07

I'm writing an ALSA SOC driver for an I2S-based device, and I'm having a really hard time understanding how ALSA uses the DMA buffers. And yes, I've read the documentation and studied some sample source code. I used to write audio drivers for a living, but that was many years ago, and it wasn't for Linux. Perhaps the concepts in my head are outdated, but I just don't see enough explanation as to how DMA buffers are supposed to work. Back then, audio drivers used "ping pong" DMA buffers. A single DMA buffer is allocated, and the audio hardware is programmed to read from that buffer in a loop. That is, it would automatically restart reading from the beginning of the buffer without any reprogramming. The hardware would also be programmed to issue an interrupt when it got to the end of the buffer, and when it got to the half-way point. To start playback, you first filled the whole buffer with audio data, and then told the hardware to start playing. After the hardware got to the half-way point, it would issue an interrupt. You would then tell the OS you need more data, and you'd get it. You then copy that data into the first half of the buffer *while* the hardware was playing the second half. Later, the hardware would interrupt you when it got to the end of the buffer. You'd then copy more data to the 2nd half while the hardware is playing the first half. And so - hardware plays one half while you copy data to the other half. Hence, "ping pong". So how do I implement this in ALSA? The "Writing an ALSA Driver" document doesn't even contain the words "ping" or "pong". -- Timur Tabi Linux Kernel Developer @ Freescale

4 6

[alsa-devel] [ASoC] bug in soc_pcm_open
by Matthieu CASTET 12 Jun '07

12 Jun '07

Hi, it seems there is a bug in soc_pcm_open. In some case [1] we are in a error path, but we don't set the ret return value. This cause bug/crash from alsa core. Thanks Matthieu [1]http://hg-mirror.alsa-project.org/alsa-kernel/file/38926de0e79c/soc/soc-c… if (!runtime->hw.rates) { 218 printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", 219 codec_dai->name, cpu_dai->name); 220 goto machine_err; 221 } 222 if (!runtime->hw.formats) { 223 printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", 224 codec_dai->name, cpu_dai->name); 225 goto machine_err; 226 } 227 if (!runtime->hw.channels_min || !runtime->hw.channels_max) { 228 printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", 229 codec_dai->name, cpu_dai->name); 230 goto machine_err; 231 }

4 4

[alsa-devel] Realtek 268
by Steven Zakulec 12 Jun '07

12 Jun '07

I have tried the current (from about 6 hours ago) drivers+lib from the HG repository. My card is now labelled as an ALC 268, and shows me what appears to be a correct number of options in the mixer. Unfortunately, I still get no sound. I've compiled the driver with --debug=full, and the lib with the debug. The following is from dmesg when I removed the old snd-hda-intel module, did a make install, then did modprobe snd-hda-intel: (I cleared out dmesg before I did the modprobe) [ 1473.892000] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 21 [ 1473.892000] PCI: Setting latency timer of device 0000:00:1b.0 to 64 [ 1474.092000] hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 1474.864000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 1474.872000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) My /proc/asound/card0/codeco#0 is as follows: cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff10 Revision Id: 0x100003 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30940: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4015812d: [N/A] Speaker at Ext N/A Conn = Optical, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x81 0x81] Connection: 7 0x18 0x19 0x1a 0x1c 0x14 0x15* 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x81 0x81] Connection: 7 0x18 0x19 0x1a 0x1c 0x14* 0x15 0x13 Output for the audio device (lspci -vvv) below: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Toshiba America Info Systems Unknown device ff10 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 21 Region 0: Memory at f0b40000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0

2 1

[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] Source of timing for MIDI and/or audio synchronization
by Mark Rivera 12 Jun '07

12 Jun '07

Hello (new to list), and thank you for the work you have been doing. As an end-user of the ALSA project's development, I am very grateful for being able to use softwares which provide audio functionality to Linux. I have been reading up on the docs and list archives, but I still have an open question for which I hope you can lead me to an answer. I designed and built a MIDI sequencer based around a Microchip PIC18 MCU, and used the built-in hardware timers and interrupt levels to drive the 'scheduler' for initiating MIDI output (EUSART) routines. In this same vein, I would like to write a software sequencer with as 'tight' timing as possible. So, in a similar concept for Linux - from where does the timing of audio and MIDI events come from? It occurred to me that it would be nice to have a loadable kernel module (LKM) to which userland programs could subscribe for receiving 'soft interrupts' based on some timer (cpu ticks?) so that various software could all reference the same notion of the "current time" - a sort of "master clock" with start/stop/continue as well. Does ALSA provide this functionality? Can someone point me in the right direction for learning more about this? Many thanks. Mark

3 2

[alsa-devel] snd_pcm_lib_malloc_pages()
by Timur Tabi 12 Jun '07

12 Jun '07

(resending since I haven't gotten a reply yet) Liam Why do none of the ASoC drivers use snd_pcm_lib_malloc_pages()? They all allocate their DMA buffers by hand. -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] [PATCH] snd-usb-caiaq: Fix compatibility with kernels < 2.6.19
by Sergey Vlasov 12 Jun '07

12 Jun '07

Kernels before 2.6.19 used a different signature for the urb->complete callback (an additional "struct pt_regs *regs" parameter was present), therefore snd-usb-caiaq needs some compatibility patches in alsa-driver, like snd-usb-audio. Signed-off-by: Sergey Vlasov <vsu(a)altlinux.ru> --- The snd_usb_caiaq_midi_output_done() part is somewhat hackish - the problem is that snd_usb_caiaq_midi_output_done() is declared in caiaq-midi.h, and the alsa-driver build system does not support patching of *.h files. The rest is basically copied from usbaudio.patch. Without the patch compiling for 2.6.18 produces warnings: .../usb/caiaq/caiaq-device.c: In function 'init_card': .../usb/caiaq/caiaq-device.c:316: warning: passing argument 6 of 'usb_fill_bulk_urb' from incompatible pointer type .../usb/caiaq/caiaq-device.c:321: warning: passing argument 6 of 'usb_fill_bulk_urb' from incompatible pointer type .../usb/caiaq/caiaq-audio.c: In function 'alloc_urbs': .../usb/caiaq/caiaq-audio.c:583: warning: assignment from incompatible pointer type --- alsa-driver-1.0.14/usb/caiaq/Makefile.alt-caiaq-api 2007-06-11 16:29:28 +0400 +++ alsa-driver-1.0.14/usb/caiaq/Makefile 2007-06-11 19:50:39 +0400 @@ -5,6 +5,13 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf +clean-files := caiaq-audio.c caiaq-device.c + include $(SND_TOPDIR)/alsa-kernel/usb/caiaq/Makefile +EXTRA_CFLAGS += -I$(SND_TOPDIR)/alsa-kernel/usb/caiaq + include $(SND_TOPDIR)/Rules.make + +caiaq-audio.c: caiaq-audio.patch $(SND_TOPDIR)/alsa-kernel/usb/caiaq/caiaq-audio.c +caiaq-device.c: caiaq-device.patch $(SND_TOPDIR)/alsa-kernel/usb/caiaq/caiaq-device.c --- alsa-driver-1.0.14/usb/caiaq/caiaq-audio.patch.alt-caiaq-api 2007-06-11 19:18:40 +0400 +++ alsa-driver-1.0.14/usb/caiaq/caiaq-audio.patch 2007-06-11 19:18:17 +0400 @@ -0,0 +1,26 @@ +--- ../../alsa-kernel/usb/caiaq/caiaq-audio.c 2007-06-11 16:29:28 +0400 ++++ caiaq-audio.c 2007-06-11 19:17:38 +0400 +@@ -464,7 +464,11 @@ static void fill_out_urb(struct snd_usb_ + check_for_elapsed_periods(dev, dev->sub_playback); + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void read_completed(struct urb *urb, struct pt_regs *regs) ++#else + static void read_completed(struct urb *urb) ++#endif + { + struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaqdev *dev; +@@ -518,7 +522,11 @@ static void read_completed(struct urb *u + usb_submit_urb(urb, GFP_ATOMIC); + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void write_completed(struct urb *urb, struct pt_regs *regs) ++#else + static void write_completed(struct urb *urb) ++#endif + { + struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaqdev *dev = info->dev; --- alsa-driver-1.0.14/usb/caiaq/caiaq-device.patch.alt-caiaq-api 2007-06-11 19:55:17 +0400 +++ alsa-driver-1.0.14/usb/caiaq/caiaq-device.patch 2007-06-11 19:54:49 +0400 @@ -0,0 +1,31 @@ +--- ../../alsa-kernel/usb/caiaq/caiaq-device.c 2007-06-11 16:29:28 +0400 ++++ caiaq-device.c 2007-06-11 19:54:46 +0400 +@@ -100,7 +100,11 @@ static struct usb_device_id snd_usb_id_t + { /* terminator */ } + }; + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void usb_ep1_command_reply_dispatch (struct urb* urb, struct pt_regs *regs) ++#else + static void usb_ep1_command_reply_dispatch (struct urb* urb) ++#endif + { + int ret; + struct snd_usb_caiaqdev *dev = urb->context; +@@ -295,6 +299,16 @@ static struct snd_card* create_card(stru + return card; + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void snd_usb_caiaq_midi_output_done_wrapper(struct urb* urb, ++ struct pt_regs *regs) ++{ ++ snd_usb_caiaq_midi_output_done(urb); ++} ++#undef snd_usb_caiaq_midi_output_done ++#define snd_usb_caiaq_midi_output_done snd_usb_caiaq_midi_output_done_wrapper ++#endif ++ + static int init_card(struct snd_usb_caiaqdev *dev) + { + char *c;

2 1

[alsa-devel] [PATCH] asihpi: Fix compatibility with 64-bit resource_size_t
by Sergey Vlasov 12 Jun '07

12 Jun '07

In recent kernels the resource_size_t type can be either 32-bit or 64-bit depending on config options. The proper way to printk such values is to cast them to unsigned long long. Signed-off-by: Sergey Vlasov <vsu(a)altlinux.ru> --- Without the patch compilation on 2.6.18 produces warnings: .../pci/asihpi/hpimod.c: In function 'adapter_probe': .../pci/asihpi/hpimod.c:460: warning: format '%x' expects type 'unsigned int', but argument 4 has type 'resource_size_t' .../pci/asihpi/hpimod.c:460: warning: format '%x' expects type 'unsigned int', but argument 5 has type 'resource_size_t' --- alsa-driver-1.0.14/pci/asihpi/hpimod.c.alt-asihpi-warnings 2007-06-11 16:29:28 +0400 +++ alsa-driver-1.0.14/pci/asihpi/hpimod.c 2007-06-11 19:58:30 +0400 @@ -457,10 +457,10 @@ static int __devinit adapter_probe(struc // fill in HPI_PCI information from kernel provided information for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { - HPI_DEBUG_LOG4(DEBUG, "Resource %d %s %x-%x\n", idx, + HPI_DEBUG_LOG4(DEBUG, "Resource %d %s %llx-%llx\n", idx, pci_dev->resource[idx].name, - pci_resource_start(pci_dev, idx), - pci_resource_end(pci_dev, idx)); + (unsigned long long)pci_resource_start(pci_dev, idx), + (unsigned long long)pci_resource_end(pci_dev, idx)); memlen = pci_resource_len(pci_dev, idx); if (memlen) {

2 1

[alsa-devel] Shutdown() vector in ALSA
by Nobin Mathew 12 Jun '07

12 Jun '07

why no ALSA driver has shutdown() vector implemented? Is this vector needed to ensure the proper shutdown of the system? What all needs to be taken care if i want to implement shutdown(), except hardware shutdown. struct snd_shutdown_f_ops { struct file_operations f_ops; struct snd_shutdown_f_ops *next; }; what this operations means? defined in sound/core/init.c Is there any need to call these functions? snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); for (i = 0; i < chip->pcm_devs; i++) snd_pcm_suspend_all(chip->pcm[i]);

3 8

Re: [alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Andrew Callan 12 Jun '07

12 Jun '07

I have been without sound using Fedora Core ever since the 2.6.16 or 2.6.17 kernel update (2.6.16 crashed during initialization, so I could not tell whether sound worked or not). It had worked well through the 2.6.15 Fedora kernels. I finally sat down tonight to work the problem as hard as I could and happened upon Eric and Tobin's timely exchange. First, simply adding probe_mask=1 to the snd-hda-intel options worked for me as well; I did not end up having to specify a specific model. Second, I too was seeing a LONG hang when trying to do: cat /proc/asound/card0/codec#0 but now with the probe_mask=1, I do not see this. If it helps, the alsa-info.sh output for my system is at http://pastebin.ca/496293 If it makes any difference, I'm not sure which model I had specified on the modprobe line at the time I captured this output. One thing I do notice is that after adding probe_mask=1, aplay -l does not show the Si3054 Modem which shows up in the alsa-info output. Thanks for the help you gave in the thread with Eric. Andrew

5 16

[alsa-devel] hda-codec - Add HP Pavillion quirk to Realtek code.
by Tobin Davis 12 Jun '07

12 Jun '07

This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Resent from the correct email address. Ignore previous email.

2 1

[alsa-devel] snd_pcm_lib_malloc_pages()
by Timur Tabi 12 Jun '07

12 Jun '07

Liam Why do none of the ASoC drivers use snd_pcm_lib_malloc_pages()? They all allocate their DMA buffers by hand. -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] [ASoC] machine driver and channel configuration
by Matthieu CASTET 12 Jun '07

12 Jun '07

Hi, If the platform/cpu_dai driver and codec support capture and playback channels. But on the board only the playback link is connected. In this configuration, the codec and platform/cpu_dai driver shouldn't try to use the capture path (and the resource associated to this). Is there a way to do something like that ? Thanks Matthieu

3 3

[alsa-devel] Questions on usb audio development
by LCID Fire 12 Jun '07

12 Jun '07

Hi. I know there is quite some documentation how to develop a pci driver for alsa - but since I'll try to get a usb device working I was wondering how to write a usb driver? Is it just the same? How to interface the various aspects of the device with regards to descriptor reading, ...

4 11

[alsa-devel] hda-codec - Add HP Pavillion quirk to Realtek code.
by Tobin Davis 11 Jun '07

11 Jun '07

This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

1 0

[alsa-devel] Patch - Revo7.1 PCM Playback Switch
by dustin＠seznam.cz 11 Jun '07

11 Jun '07

Hello Takashi, I have attached a patch to support mute for front channels of M-Audio Revolution 7.1 (the DAC AK4381 features a mute bit). Signed-off-by: Pavel Hofman <dustin at seznam.cz> Please could I ask you for your opinion on the following issue (so far no response): I am working together with Ctirad on supporting more capture channels in E-MU1212. So far we have managed to increase the number of channels from the default of 8 to 16 (A_FXWC2 = 0xffffffff, added new snd_emu10k1_audigy_dsp_convert_32_to_2x16 calls in emufx.c). There is still a number of issues to solve but the 16 channel capture already works fine. I would like to get your opinion on the form of adding this support to the module. The options are either hard-coding the 16 capture channels into the code, assuming that users would prefer as many channels as possible and do the routing in jack, or introducing a module parameter emu1010-capture-channels (range e.g. 1 to 16 - untested yet) and leaving the default number at the existing 8. Furthermore, the PCMs other than efx seem to be of no use for the EMU-1x1x models, probably fairly confusing for new users. Would you find it OK, if they were removed for this specific card (if (emu->card_capabilities->emu1010)), keeping only the playback and capture PCM id 2? Thanks a lot for your suggestions. Based on them, I will prepare the patch (with comments on what we found out about the card - it is a pitty that the driver has almost no big-picture comments on the supported cards' specifics, making new developers spend a number of long nights going through the nontrivial code and guessing). Best regards, Pavel.

4 7

Re: [alsa-devel] [2/3] 2.6.22-rc4: known regressions v2
by Michal Piotrowski 09 Jun '07

09 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc4. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown IDE Subject : libata and legacy ide pcmcia failure References : http://lkml.org/lkml/2007/5/17/305 Submitter : Robert de Rooy <robert.de.rooy(a)gmail.com> Handled-By : Mark Lord <mlord(a)pobox.com> Status : problem is being debugged SATA/PATA Subject : 22-rc3 broke the CDROM in Dell notebook References : http://lkml.org/lkml/2007/5/27/63 Submitter : Gregor Jasny <gjasny(a)googlemail.com> Handled-By : Tejun Heo <htejun(a)gmail.com> Jeff Garzik <jeff(a)garzik.org> Caused-By : Tejun Heo <htejun(a)gmail.com> commit d4b2bab4f26345ea1803feb23ea92fbe3f6b77bc Status : problem is being debugged Sparc64 Subject : 2.6.22-rc broke X on Ultra5 References : http://lkml.org/lkml/2007/5/22/78 Submitter : Mikael Pettersson <mikpe(a)it.uu.se> Handled-By : David Miller <davem(a)davemloft.net> Status : problem is being debugged Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

2 1

[alsa-devel] probing state of high definition audio card in windows / SigmaTel STAC9221 microphone problems
by Eran Guendelman 09 Jun '07

09 Jun '07

Hi, I am struggling to get the (external) microphone working on a SigmaTel STAC9221 A1 device on my Dell XPS M1210 laptop. I saw other emails indicating that this was still problematic, and the latest drivers from alsa 1.0.14 did not seem to fix the problem. So now that I've started reading and trying to understand the basics behind the hda codec in an effort to fix this problem, I was wondering if it was possible for me to boot into windows (I've got a dual-boot setup), get into a state where the microphone is working, and somehow probe the codec's state at that time (all of the pin configurations, etc.)... then theoretically, if I reproduce that configuration on the linux side, it should work (right?). So the question is, does anyone have a ready-made utility for Windows to do this probe of the high definition audio codec? I could probably try writing one myself, but thought I'd check if it already exists. Thanks. Please send responses by email too (not subscribed).

3 3

[alsa-devel] Sound Devices USBPre
by Brian Phelps 08 Jun '07

08 Jun '07

Hey guys, I contacted Sound Devices at sounddevices.com about writing a Linux driver (I was going to head the project but would take any help I can get) for the USBPre, a 2ch-in 2ch-out USB preamp with highquality balanced XLR Mic inputs. They said that they would love to have a Linux Driver and will provide specs for me, I am still waiting on the info though. I have not asked if I can view their firmware sourcecode yet, and if it requires an NDA, I will have to work around that. What advice do you have for writing a driver for alsa? I have written some small kernel drivers for embedded Linux devices, a few audio programs in C, and I come from a DSP background, but I am not sure exactly where to start, besides the specs and firmware. Thanks for your help -Brian Phelps

1 0

[alsa-devel] Callback trouble
by Juan Manuel Lopez Baio 08 Jun '07

08 Jun '07

Hello. I'm having some problems with the asynchronous usage of ALSA API, and I'm not sure if this usage is correct. What I have done so far is the following: I have an class that encapsulates the ALSA API, working in a multithreaded environment. I use alsa to capture audio and want to do so in an asynchronous and non-blocing way, so I defined a static method in my class that functions as callback, which receives a pointer to the instance of my class and calls on a member function, like this: void alsaDevice::myCallback(snd_async_handler_t *handler) //STATIC { alsaDevice *dev = (alsaDevice*)snd_async_handler_get_callback_private(handler); dev->triggerCondition(); } void alsaDevice::triggerCondition() { LogError(0,"Trigger en placa %d device %d",cn,dn); m_condMutex.lock(); m_noDataCondition.trigger(); m_condMutex.unlock(); } To init ALSA, I do this: int mode = SND_PCM_ASYNC | SND_PCM_NONBLOCK; if ((err=snd_pcm_open(&pcm_cap_handle, pcm_name.str().c_str(), SND_PCM_STREAM_CAPTURE, mode) < 0)) { LogError(0,"Opening ALSA PCM device %s: %s",pcm_name.str().c_str(),snd_strerror(err)); pcm_cap_handle=NULL; return false; } then set all the pertinent parameters, and finally if((asyncaddresult=snd_async_add_pcm_handler(&ahandler,pcm_cap_handle,alsaDevice::myCallback,this))<0) { LogError(0,"No se pudo registrar la callback; PCM Error: %i",asyncaddresult); } if ((asyncaddresult=snd_pcm_start(pcm_cap_handle))<0) LogError(0,"No se pudo start pcm; PCM Error: %i",asyncaddresult); Afterwards, the first time that a call to snd_pcm_readn returns EAGAIN I call wait() on a condition object with a 100 ms timeout, which is triggered by the callback, as seen before. The problem is that the callback never seems to get called (so shows the log), so I'm not really sure if the previous steps were right. Any light you can shed on this would be much appreciated. Thanks! juan

1 0

[alsa-devel] snd-usb-audio: Add another QuickCam ID
by Ritesh Raj Sarraf 08 Jun '07

08 Jun '07

Hi, This patch adds another Logitech QuickCam USB ID for the device which is shipped with Dell's XPS M1210 Notebooks. Signed-off-by: Ritesh Raj Sarraf <rrs(a)researchut.com> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 374fbf6..c69a56f 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -70,6 +70,15 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08c6, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, PS: Request you to CC me if there are any comments. I'm not subscribed to the list. Thanks, Ritesh -- Ritesh Raj Sarraf RESEARCHUT - http://www.researchut.com "Necessity is the mother of invention."

2 2

[alsa-devel] [PATCH] soc/sh: let SND_SOC_PCM_SH7760 depend on SH_DMABRG
by Manuel Lauss 08 Jun '07

08 Jun '07

Let SND_SOC_PCM_SH7760 depend on SH_DMABRG rather than select'ing it. Gets rid of the Kbuild warning about SH_DMABRG being undefined on non-sh builds. Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- linux-2.6.22-rc4-mm2/sound/soc/sh/Kconfig~ 2007-06-07 11:09:30.000000000 +0200 +++ linux-2.6.22-rc4-mm2/sound/soc/sh/Kconfig 2007-06-07 11:09:30.000000000 +0200 @@ -2,8 +2,7 @@ menu "SoC Audio support for SuperH" config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" - depends on CPU_SUBTYPE_SH7760 && SND_SOC - select SH_DMABRG + depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG help Enable this option for SH7760 AC97/I2S audio support.

2 1

[alsa-devel] [PATCH] Add quirk for another MSI laptop to alc883_cfg_tbl[]
by Daniel T. Chen 08 Jun '07

08 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip adding the SSID of another MSI laptop to the quirk list. Thanks, Daniel Chen

2 1

[alsa-devel] [ASoC]one Small Suggestion
by Nobin Mathew 08 Jun '07

08 Jun '07

Liam, Thank you for your support, my work is almost complete now, driver is working perfectly now. I have one suggestion, I modified the wm9713.c codec driver in probe function , so that if(socdev->machine->dai_link->cpu_dai->private_data) codec->ac97->bus->private_data = socdev->machine->dai_link->cpu_dai->private_data; Here i am assigning the controller private data. so that my following ac97 read and write to access codec registers become good. static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct platform_controller *controller = ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct platform_controller *controller = ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; u32 v; Is there a way to accomodate this in existing ASoC design? Or a better way to do this?

3 5

[alsa-devel] [RFC] [PATCH 1/2] ASoC sound support for SMDK2440 boards: UDA1341 driver
by Zoltan Devai 08 Jun '07

08 Jun '07

ASoC Support for the UDA1341 codec. Please note that until there is a final L3-bus driver, this patch won't work. --- --- /dev/null +++ /sound/soc/codecs/uda1341.c @@ -0,0 +1,519 @@ +/* + * uda1341.c -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/l3.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "../s3c24xx/s3c24xx-i2s.h" +#include "uda1341.h" + +#define UDA1341_DEBUG 0 + +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \ + printk(KERN_DEBUG "UDA1341: %s:" format "\n" ,__FUNCTION__, ## arg) +#else +#define DBG(format, arg...) do {} while (0) +#endif + +#define UDA1341_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA1341_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE ) + +struct uda1341_priv { + int sysclk; + int dai_fmt; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda1341_reg[UDA1341_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda1341_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + DBG("reg: %d", reg); + + if (reg >= UDA1341_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda1341_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + DBG("reg: %02X", reg); + + if (reg >= UDA1341_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda1341 registers + * + */ +static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + int addr; + u8 data = value; + + DBG("reg: %02X, value:%02X", reg, value); + + if (reg >= UDA1341_REGS_NUM) { + DBG("Unkown register: reg: %d", reg); + return -EINVAL; + } + + uda1341_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA1341_STATUS0: + case UDA1341_STATUS1: + addr = UDA1341_STATUS_ADDR; + break; + case UDA1341_DATA000: + case UDA1341_DATA001: + case UDA1341_DATA010: + addr = UDA1341_DATA0_ADDR; + break; + case UDA1341_DATA1: + addr = UDA1341_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA1341_EXTADDR_PREFIX); + + ret = codec->hw_write_l3(codec->control_data, + UDA1341_DATA0_ADDR, (char*)&addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA1341_DATA0_ADDR; + data = (value | UDA1341_EXTDATA_PREFIX); + break; + } + + ret = codec->hw_write_l3(codec->control_data, addr, (char*)&data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda1341_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + uda1341_write(codec, UDA1341_STATUS0, reset_reg | (1<<6)); +} + +static int uda1341_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda1341_read_reg_cache(codec, UDA1341_DATA010); + + DBG("mute: %d", mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda1341_write(codec, UDA1341_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda1341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + u8 hw_params = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + DBG("sysclk: %d, rate:%d", uda1341->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda1341->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + return -EINVAL; + break; + } + + DBG("dai_fmt: %d, params_format:%d", uda1341->dai_fmt, + params_format(params)); + + + /* set DAI format and word length */ + switch (uda1341->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch(params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + return -EINVAL; + } + + uda1341_write(codec, UDA1341_STATUS0, hw_params); + + return 0; +} + +static int uda1341_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("clk_id: %d, freq: %d, dir: %d", clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + we'll error out on set_hw_params if it's not OK */ + if ( (freq >= (256 * 8000)) && (freq <= (512 * 48000)) ) { + uda1341->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int uda1341_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("fmt: %08X", fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda1341->dai_fmt = fmt; + + return 0; +} + +static int uda1341_dapm_event(struct snd_soc_codec *codec, int event) +{ + u8 reg; + + DBG("event: %08X", event); + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* ADC, DAC on */ + reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg | 0x03); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* mute, ADC, DAC power off */ + reg = codec->read(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg & ~(0x03)); + break; + } + codec->dapm_state = event; + return 0; +} + +static const char *uda1341_dsp_setting[] = {"Flat", "Minimum1", "Minimum2", "Maximum"}; +static const char *uda1341_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda1341_mixmode[] = {"DD", "Input 2", "Input 2", "Digital mixer"}; + +static const struct soc_enum uda1341_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA1341_DATA010, 0, 0x03, uda1341_dsp_setting), +SOC_ENUM_SINGLE(UDA1341_DATA010, 3, 0x03, uda1341_deemph), +SOC_ENUM_SINGLE(UDA1341_EA010, 0, 0x03, uda1341_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Playback Volume", UDA1341_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Mic gain", UDA1341_EA010, 2, 0x07, 0), +SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1), +SOC_SINGLE("Input channel 2 amp gain", UDA1341_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Bass boost", UDA1341_DATA001, 2, 0xF, 0), +SOC_SINGLE("Treble", UDA1341_DATA001, 0, 3, 0), + +SOC_ENUM("DSP setting", uda1341_mixer_enum[0]), +SOC_ENUM("Playback De-emphasis", uda1341_mixer_enum[1]), +SOC_ENUM("Mixer mode", uda1341_mixer_enum[2]), + +/* This should be an ext control with own handler, if one wants + to set the values in 0.5dB steps instead of 3dB */ +SOC_SINGLE("AGC output level", UDA1341_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC time const", UDA1341_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity switch", UDA1341_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity switch", UDA1341_STATUS1, 3, 1, 0), +SOC_SINGLE("Double speed playback switch", UDA1341_STATUS1, 2, 1, 0), +}; + +static int uda1341_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1341_snd_controls); i++) { + if ((err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1341_snd_controls[i],codec, NULL))) < 0) + return err; + } + + return 0; +} + +struct snd_soc_codec_dai uda1341_dai = { + .name = "UDA1341", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* pcm operations */ + .ops = { + .hw_params = uda1341_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda1341_mute, + .set_sysclk = uda1341_set_dai_sysclk, + .set_fmt = uda1341_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda1341_dai); + + +static int uda1341_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda1341_priv *uda1341; + void * codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + + printk(KERN_INFO "UDA1341 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda1341 = kzalloc(sizeof(struct uda1341_priv), GFP_KERNEL); + if (uda1341 == NULL) + goto priv_err; + + codec->private_data = uda1341; + + codec->reg_cache = kmemdup(uda1341_reg, sizeof(uda1341_reg), \ + GFP_KERNEL); + + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda1341_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA1341"; + codec->owner = THIS_MODULE; + codec->dai = &uda1341_dai; + codec->num_dai = 1; + codec->read = uda1341_read_reg_cache; + codec->write = uda1341_write; + codec->dapm_event = uda1341_dapm_event; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA1341 SoC codec: \ + Unable to find an L3 bus adapter\n"); + ret = -ENODEV; + goto pcm_err; + } + + codec->control_data = codec_setup_data; + codec->hw_write_l3 = (hw_write_l3_t)l3_write; + + uda1341_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda1341_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register controls\n"); + goto pcm_err; + } + + //uda1341_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda1341_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda1341_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int uda1341_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1341_reg); i++) { + codec->write(codec, i, *cache++); + } + uda1341_dapm_event(codec, codec->suspend_dapm_state); + return 0; +} +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda1341 = { + .probe = uda1341_soc_probe, + .remove = uda1341_soc_remove, +#if defined(CONFIG_PM) + .suspend = uda1341_soc_suspend, + .resume = uda1341_soc_resume, +#endif /* CONFIG_PM */ +}; + +EXPORT_SYMBOL(soc_codec_dev_uda1341); + +MODULE_DESCRIPTION("UDA1341 ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai"); +MODULE_LICENSE("GPL"); --- /dev/null +++ /sound/soc/codecs/uda1341.h @@ -0,0 +1,47 @@ +/* + * uda1341.h -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _UDA1341_H +#define _UDA1341_H + +#define UDA1341_L3ADDR 5 +#define UDA1341_DATA0_ADDR ((UDA1341_L3ADDR << 2) | 0) +#define UDA1341_DATA1_ADDR ((UDA1341_L3ADDR << 2) | 1) +#define UDA1341_STATUS_ADDR ((UDA1341_L3ADDR << 2) | 2) + +#define UDA1341_EXTADDR_PREFIX 0xC0 +#define UDA1341_EXTDATA_PREFIX 0xE0 + +/* UDA1341 registers */ +#define UDA1341_EA000 0 +#define UDA1341_EA001 1 +#define UDA1341_EA010 2 +#define UDA1341_EA011 3 +#define UDA1341_EA100 4 +#define UDA1341_EA101 5 +#define UDA1341_EA110 6 +#define UDA1341_EA111 7 +#define UDA1341_STATUS0 8 +#define UDA1341_STATUS1 9 +#define UDA1341_DATA000 10 +#define UDA1341_DATA001 11 +#define UDA1341_DATA010 12 +#define UDA1341_DATA1 13 + +#define UDA1341_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_codec_dai uda1341_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda1341; + +#endif /* _UDA1341_H */

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[alsa-devel] [PATCH] Add SSID for Dell XPS400 to stac922x_cfg_tbl[]
by Daniel T. Chen 08 Jun '07

08 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip adding a quirk entry for a Dell XPS400. Thanks, Daniel Chen

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[alsa-devel] Testing/forcing XRUN
by Eliot Blennerhassett 08 Jun '07

08 Jun '07

Hi, I want to test my driver behaviour when an xrun happens. I have tried using aplay/arecord, and suspending (ctrl-Z) them while playing/recording but this doesn't have the desired effect. (I don't see my driver code detecting an xrun). Instead I see a trigger-stop command to the driver. The apps do report an overrun when they are restarted though. Can anyone answer these: 1) Is this a valid way to simulate an xrun due to application not keeping up? 1.1) If not, what is? Do I have to write an app that deliberately starves the streams? 2) Where is the xrun being detected if my driver doesnt see it? Where does the trigger-stop command come from when I suspend the app? thanks Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] Audio driver for omap 24xx.
by Pharaoh . 07 Jun '07

07 Jun '07

Hi I am referring the EAC driver for omap 24xx which is used by n800 board. The EAC 24xx driver doesn't look complete since it has only hardware init sequences and very basic ALSA related functionalities i.e. register the sound card etc. Is omap 24xx eac driver still in intermediate stage?Or it is complete but not yet integrated in to omap git tree? In later case, where can I find the complete eac driver? Also, omap 24xx driver doesn't seem to follow the framework/structure of other omap alsa drivers in sound/arm/omap/ , any particular reason? Please enlighten. I know this might be little off topic here, but there are only two places where I can ask this question, so I am trying both. -pharaoh.

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[alsa-devel] Surround sound problem with CS46xx
by Vasilis Manwlopoulos 06 Jun '07

06 Jun '07

Hi to everybody, I am kind of new in linux and to alsa and I am facing a problem with the setup for 5.1 surround sound with my sound card Hercules Fortissimo III, recognized having the chip cs46xx. I have managed to enable surround sound by creating a .asounrc file with the followings pcm.!default { type plug slave.pcm "surround51" slave.channels 6 route_policy duplicate } The problem is that I am facing weird cpu load spikes (sometimes isn't just spikes, but just 99% usage for a long time!) even when just playing a normal mp3 file at 128Kbps. This is happening when playing with various players (totem,rhythbox,vlc). Also the sound has cracklings or there is a delay between the front and the rear speakers. i have send this email to alsa-users too, but since there wasn't anyone to propose a workaround I was adviced to also send it to alsa-devel. I am using the distro ubuntu 7.04 with the default alsa libraries version 1.0.13ubuntu5. So, if anyone has any idea plz give me a hint, cause I am stuck with it about two weeks now, without having any clue to proceed with!!! PS. If any other information or command output is necessary to help me, tell me so.

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[alsa-devel] [ASoC]Changing the order of trigger in soc_pcm_trigger
by Nobin Mathew 06 Jun '07

06 Jun '07

Liam, I have one suggestion. Can we use this if (rtd->codec_dai->ops.trigger) { ret = rtd->codec_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (rtd->cpu_dai->ops.trigger) { ret = rtd->cpu_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (platform->pcm_ops->trigger) { ret = platform->pcm_ops->trigger(substream, cmd); if (ret < 0) return ret; } Instead of if (rtd->codec_dai->ops.trigger) { ret = rtd->codec_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (platform->pcm_ops->trigger) { ret = platform->pcm_ops->trigger(substream, cmd); if (ret < 0) return ret; } if (rtd->cpu_dai->ops.trigger) { ret = rtd->cpu_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } That means trigger the Cpu before pcm. Because most of the controllers needs to be initialized and triggered before external DMA is triggered. Correct me if i am going wrong. This was causing so many problems in my hardware. Nobin Mathew

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[alsa-devel] Realtek 268
by Steven Zakulec 06 Jun '07

06 Jun '07

Hello ALSA developers, I have one of the Intel HDA cards that is not supported currently: Realtek 268. It was previously mentioned in May (http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/000828.html). The post seemed to imply it was not a terribly difficult task to add support for the card, but was very vague on details. If it is in fact just a matter of copying an existing preset and changing the values, I would probably try to do it myself. If it is any more difficult than that, I'd like to formally ask the developers if they could add any support for the card. The lspci and /proc/asound/codec info is identical to that in the posting I mentioned. If you need anymore information, I would be happy to provide it. Thank you!

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[alsa-devel] Connecting two USB-Soundcard to JACK
by Carsten Beth 06 Jun '07

06 Jun '07

Hello! We want to connect two USB soundcards to one multichannel device at the alsa layer and to use that virtual device with JACK. But we don't find a way to do so. The USB soundcard is self-made. It woks as a high speed USB device with sample rates of up to 96 KHz, has four capture and two playback cannels. The clocks of two cards can be synchronized at hardware level. Using one card by its own works perfectly both, at the alsa and at the JACK level, also as a duplex device. All is done with alsalib 1.0.13, jackd 0.101.1 and qjackctl 0.2.21. Now I'm trying to build a virtual device with 8 capture and 4 playback channels at the alsa level. But, as it can be read at several locations, this only works for capture only or playback. It has been reported, that duplex operation is possible after patching pcm_multi.c. I have applyed the patch, but this has not changed anything. Now I want to ask you for support, to get the device running. I made some experiments with different configurations of alsa/jackd and had some strange results. You can read about the results and used the asoundrc below. Questions, that I can't answer are - Why do I see just the channels of one card when using the multi plugin in try 2), 5) and 7)? - Why is the processor load so high with playback only in try 2) and 3) but not in duplex operation? - Why are there lots non-adible xruns with playback only in try 2) and 3)? 1) playback only with p_multi ALSA: mmap-based access is not possible for the playback stream of this audio interface 2) playback only with p_multi_plug Playback of just the 1st cards two channel. Lot of processor load: Both cores of the Core Duo 2GHz system are at about 90% load. Lot of XRUNs: 08:01:35.514 XRUN callback (48501 skipped). 08:01:37.517 XRUN callback (80633 skipped). 08:01:39.519 XRUN callback (70612 skipped). but XRUNS can't be heard. Sometimes jackd terminates after a while: ALSA: could not complete playback of 256 frames: error = -32 cycle execution failure, exiting DRIVER NT: could not run driver cycle jack main caught signal 12 [...] zombified - calling shutdown handler [...] 08:01:40.926 Post-shutdown script terminated with exit status=256. 3) playback only with p_multi_route Number of channels as expected (4 channels, two from each card). Lot of processor load, like 2) Lot of non-adible XRUNS, like 2). Sometimes jackd terminates after a while with the same message as in 2). 4) capture only with c_multi ALSA: mmap-based access is not possible for the capture stream of this audio interface 5) capture only with c_multi_plug Captures just two channel from the 1st card Low processor load (near 0%) No XRUNS. 6) capture only with c_multi_route Number of channels as expected (8 channels, four from each card). Low load and no XRUNS, like 5) 7) duplex with p_multi_plug and c_multi_plug Playback and capture works just for the 1st card (two playback and 4 capture channels). Low processor load, like 2) Lot of XRUNs, but less the 2) (and now the XRUNS can be heard). Here is a snip: **** alsa_pcm: xrun of at least 0.010 msecs 11:34:00.013 XRUN callback (9 skipped). **** alsa_pcm: xrun of at least 0.014 msecs 11:34:02.037 XRUN callback (11 skipped). **** alsa_pcm: xrun of at least 0.018 msecs **** alsa_pcm: xrun of at least 0.016 msecs 11:34:04.069 XRUN callback (11 skipped). **** alsa_pcm: xrun of at least 0.016 msecs 11:34:06.099 XRUN callback (11 skipped). **** alsa_pcm: xrun of at least 0.018 msecs **** alsa_pcm: xrun of at least 0.013 msecs 11:34:08.116 XRUN callback (12 skipped). Sometimes jackd terminates after a while with the same message as in 2). 8) duplex with p_multi_route and c_multi_route Exactly the same as in 7) expect for the number of channels. Now all channels are available (four playback and four capture channels) Here the used asoundrc: ##################################### # 1st USB card ##################################### pcm.usb0 { type hw card 1 } ctl.usb0 { type hw card 1 } ##################################### # 2nd USB card ##################################### pcm.usb1 { type hw card 2 } ctl.usb1 { type hw card 2 } pcm.p_multi { type multi; slaves.a.pcm "usb0"; slaves.a.channels 2; slaves.b.pcm "usb1"; slaves.b.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1; } ctl.p_multi { type hw; card 1; } pcm.c_multi { type multi; slaves.a.pcm "usb0"; slaves.a.channels 4; slaves.b.pcm "usb1"; slaves.b.channels 4; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave a; bindings.2.channel 2; bindings.3.slave a; bindings.3.channel 3; bindings.4.slave b; bindings.4.channel 0; bindings.5.slave b; bindings.5.channel 1; bindings.6.slave b; bindings.6.channel 2; bindings.7.slave b; bindings.7.channel 3; } ctl.c_multi { type hw; card 1; } pcm.p_multi_plug { type plug; slave.pcm "p_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; } ctl.p_multi_plug { type hw; card 1; } pcm.c_multi_plug { type plug; slave.pcm "c_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; ttable.4.4 1; ttable.5.5 1; ttable.6.6 1; ttable.7.7 1; } ctl.c_multi_plug { type hw; card 1; } pcm.p_multi_route { type route; slave.pcm "p_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; } ctl.p_multi_route { type hw; card 1; } pcm.c_multi_route { type route; slave.pcm "c_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; ttable.4.4 1; ttable.5.5 1; ttable.6.6 1; ttable.7.7 1; } ctl.c_multi_route { type hw; card 1; } Thank you Carsten Beth -- Dipl.-Inform. Carsten Beth OFFIS - Institute for Information technology R&D Division Embedded Hardware-/Software-Systems Escherweg 2 - 26121 Oldenburg - Germany - http://www.offis.de/ Phone: +49-441-9722-241 - Fax: +49-441-9722-282

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Re: [alsa-devel] Power management state flow in ALSA
by Nobin Mathew 06 Jun '07

06 Jun '07

Liam, In soc_resume() we should do this snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); instead of snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); Because snd_pcm_resume in pcm_native.c expects the card to be in SNDRV_CTL_POWER_D0 state. Other ioctl will wait forever. static int snd_pcm_resume(struct snd_pcm_substream *substream) { struct snd_card *card = substream->pcm->card; int res; snd_power_lock(card); if ((res = snd_power_wait(card, SNDRV_CTL_POWER_D0)) >= 0) res = snd_pcm_action_lock_irq(&snd_pcm_action_resume, substream, 0); snd_power_unlock(card); return res; } One more thing why we are using snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3cold); every other driver is doing snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); SNDRV_CTL_POWER_D3cold is true for deep sleep where power to the controller is switched off . In sleep, only clock to the controller is switched off. Nobin Mathew On 6/4/07, Nobin Mathew <nobin.mathew(a)gmail.com> wrote: > Liam > > I will get back, i need to set up the board > > On 6/1/07, Liam Girdwood <lg(a)opensource.wolfsonmicro.com> wrote: > > Nobin, > > > > Can you try the following patch :- > > > > http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… > > > > This will notify pcm's of their suspend state. > > > > Liam > > > > On Wed, 2007-05-30 at 12:58 +0200, Takashi Iwai wrote: > > > At Wed, 30 May 2007 16:06:54 +0530, > > > Nobin Mathew wrote: > > > > > > > > I am writing an ASoC driver, where i can place these calls > > > > > > > > snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); > > > > snd_pcm_suspend_all(chip->pcm[i]); > > > > > > > > and snd_power_change_state(card, SNDRV_CTL_POWER_D0); > > > > > > > > > > > > In soc-core.c ? > > > > > > I suppose so. > > > In soc_suspend() and soc_resume(), you can get the card instance from > > > codec->card, at least. > > > > > > But currently SoC doesn't keep the PCM instances, so it cannot be > > > implemented as it is... Liam, any plan or known issue regarding this? > > > > > > > > > Takashi > > > > >

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[alsa-devel] Problem restarting alsa: can't set buffer size
by Juan Manuel Lopez Baio 06 Jun '07

06 Jun '07

Hello. I'm developing an application that records input from ALSA and process it. To do so I user the API starting with snd_pcm_open and follow with the parameters settings and so on, and once the app is done and exits, it closes with snd_pcm_hw_params_free and snd_pcm_close. The problem is that afterwards, if I restart the application, I can't set up ALSA, becausea continuous calls to snd_pcm_hw_params_set_buffer_size always fail (I start with a buffer size of 8192 and keep dividing it by 2). I finally have to restart the system in order to get it to work again. I'm on a Mandriva box with gcc 4.1.2 ¿Any thoughts on this? Thanks in advance! juan

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[alsa-devel] using 'type softvol' in a card conf file in /usr/share/alsa/cards, am i doing this right?
by John L. Utz III 05 Jun '07

05 Jun '07

Hello; I am working with an CMEDIA 8738-MC6 and i am trying to gain control of the second DAC by way of softvolume. but i seem to not be succeeding because my application cant find my 'Rear' control, am i doing this correctly? the original /usr/share/alsa/cards/CMI8738-MC6.conf CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 1 } i have modified it to look thusly: notice that the type is now softvol and the original hardware description is encapsulated in slave.pcm # 2nd DAC CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type softvol slave.pcm { type hw card $CARD device 1 } control { name "Rear Playback Volume" card $CARD } } Shouldnt i now be able to see 'Rear Playback Volume' in the output of amixer controls ?? as it stands right now, i can not. what am i doing incorrectly? tnx! johnu

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Re: [alsa-devel] [2/5] 2.6.22-rc4: known regressions
by Michal Piotrowski 05 Jun '07

05 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc4. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions ACPI Subject : ACPI_CPUFREQ fails to load - HP ML110 G4 Xeon 3040 References : http://bugzilla.kernel.org/show_bug.cgi?id=8570 Submitter : Marc Tuerschmann <marc(a)tuerschmann.de> Status : Unknown (possible hardware issues) Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown File systems Subject : OOPS (NULL pointer dereference) in v2.6.22-rc3 References : http://lkml.org/lkml/2007/6/1/389 Submitter : Alex Riesen <raa.lkml(a)gmail.com> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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[alsa-devel] Alsa-OSS Duplex bug (revisited)
by Alan Horstmann 05 Jun '07

05 Jun '07

About a year ago I posted an explaination of the cause of non-functioning with Alsa-OSS in duplex, together with a workaround patch. After some discussion the thread ended. However since then there has been a steady trickle of downloads of the patches from my host (it can also be found elsewhere) and some reports of great success,eg http://audacityteam.org/forum/thread/1388 (2nd page especially) In attempting to tidy up some loose ends, I hope you can bear with me revisiting this and making the case for it's acceptance. I am aware the Alsa-OSS is not of much interest to developers, and it might be said 'use native Alsa'; however that is not possible or desirable in all applications, and anyway, why provide an OSS emulation that cannot be made to work properly? To summarise the problem first: OSS provides a single SNDCTL_DSP_CHANNELS ioctl; where a device is duplex the number of capture and playback channels cannot be set separately. With original OSS this was not a problem AFAIK because separate devices are created for capture and playback, ie they are not used duplex. However Alsa-OSS in most cases creates a single combined device, which only operates correctly when an equal number of capture and playback channels are used. The Workaround: The workaround I proposed and now use constantly is like a secret trapdoor. Where the existing interface is adequate (ie captue channels == playback channels, or non-duplex) nothing changes, and SNDCTL_DSP_CHANNELS is set with a number between 0 and 128. It is fully compatible with apps that don't know of the secret door -the front door is used! However, if an app needs to set capture channels != playback channels, in duplex, at present the stream will malfunction every time. But with this patch applied, if the app knows the workaround, then it can simply set SNDCTL_DSP_CHANNELS as playback + 256*capture channels, and this will be interpreted correctly in the patched Alsa-OSS and the stream functions correctly. As I say, there is therefore no loss of compatibility in either direction. Providing the app only uses the special formula when necessary rather than always, it is only providing a way to fix something that is broken. It would seem to me of value to add the patch to Alsa so that the workaround can be effected by changes to the app only, without also having to patch and compile a replacement Alsa. Whilst it is by no means ideal, it does overcome the underlying problem successfully. Details of the workaround could be added to the OSS-Emulation.html document. Patch for 1.0.14rc4 is attached for consideration, together with example portaudio patch. Regards Alan

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[alsa-devel] ALSA 1.0.14 release
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hi all, finally, ALSA 1.0.14 is out. For list of changes, please, check: http://www.alsa-project.org/changes/v1-0-13--v1-0-14.txt http://www.alsa-project.org/changes/v1-0-14rc4--v1-0-14.txt Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Alsa guru needed for short-term contracting gig
by Darren Gibbs 04 Jun '07

04 Jun '07

Are there any SF Bay Area local Alsa experts out there that could provide assistance bringing up Alsa with driver for an I2S codec on an ARM embedded system?

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[alsa-devel] [PATCH] ALSA: Fix prototype for snd_soc_dai_link.init() function pointer
by Timur Tabi 04 Jun '07

04 Jun '07

Change the snd_soc_dai_link.init() function to take a pointer to the corresponding snd_soc_dai_link structure, instead of a pointer to a snd_soc_codec structure. This allows the initialization function to initialize its own structure. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This is a fix for ALSA bug 3135 (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3135). Since I don't have an ARM cross-compiler, and currently all ASoC drivers are for ARM platforms only, I'm not sure if this fix is correct. include/sound/soc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 3 ++- sound/soc/pxa/corgi.c | 3 ++- sound/soc/pxa/poodle.c | 3 ++- sound/soc/pxa/spitz.c | 3 ++- sound/soc/pxa/tosa.c | 3 ++- sound/soc/soc-core.c | 2 +- 7 files changed, 12 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index db6edba..86e1b1d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -410,7 +410,7 @@ struct snd_soc_dai_link { struct snd_soc_ops *ops; /* codec/machine specific init - e.g. add machine controls */ - int (*init)(struct snd_soc_codec *codec); + int (*init)(struct snd_soc_dai_link *codec); }; /* SoC machine */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 820a676..a4f5a99 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -216,9 +216,10 @@ static const char *intercon[][3] = { /* * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) +static int eti_b1_wm8731_init(struct snd_soc_dai_link *dai) { int i; + struct snd_soc_codec *codec = dai->codec_dai->codec; DBG("eti_b1_wm8731_init() called\n"); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5ee51a9..43237f9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -286,9 +286,10 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { /* * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device */ -static int corgi_wm8731_init(struct snd_soc_codec *codec) +static int corgi_wm8731_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 0915cf7..5e8cb76 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -246,9 +246,10 @@ static const snd_kcontrol_new_t wm8731_poodle_controls[] = { /* * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device */ -static int poodle_wm8731_init(struct snd_soc_codec *codec) +static int poodle_wm8731_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 80e8210..95dcaa4 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -292,9 +292,10 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = { /* * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device */ -static int spitz_wm8750_init(struct snd_soc_codec *codec) +static int spitz_wm8750_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; /* NC codec pins */ snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 5504e30..b438349 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -192,9 +192,10 @@ static const struct snd_kcontrol_new tosa_controls[] = { tosa_set_spk), }; -static int tosa_ac97_init(struct snd_soc_codec *codec) +static int tosa_ac97_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "OUT3", 0); snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 92d5d91..654a517 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1093,7 +1093,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); for(i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { - err = socdev->machine->dai_link[i].init(codec); + err = socdev->machine->dai_link[i].init(&machine->dai_link[i]); if (err < 0) { printk(KERN_ERR "asoc: failed to init %s\n", socdev->machine->dai_link[i].stream_name); -- 1.5.0.2.260.g2eb065

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[alsa-devel] SALSA-Lib version 0.0.1 release
by Takashi Iwai 04 Jun '07

04 Jun '07

Hi all, I created a small replacement of ALSA-library, mainly for embedded systems with limited resources. The library is named as "SALSA-Lib" (too easy, eh?) It has no plugin layer but only hw layer, thus no sexy features like dmix and format conversions are available. It supports no shiny and complex configuration files. And it supports no sequencer. The only goal of this library is the reduction of the size of library and running binaries by inlining functions with direct syscalls. The library is designed to be source-level API compatible with the normal alsa-lib 1.0.14. Note that it is _not_ designed to be binary-compatible at all, and it will be never so. The first released version 0.0.1 is available at: ftp://ftp.suse.com/pub/people/tiwai/salsa-lib/salsa-lib-0.0.1.tar.bz2 The package is released under LGPL as well as the original alsa-lib. I've tested it only on i386, and tests on other architectures would be greatly appreciated. (And, no, I didn't make it on HG tree yet.) Takashi

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[alsa-devel] Datasheet Needed
by chris＠idlelion.net 04 Jun '07

04 Jun '07

(working on bug #305) I'm looking for a proper datasheet on the Cirrus Logic / Crystal Semiconductor CS4624 (or CS4622) sound chip. It's used in a number of popular laptops made around 1999-2001. I don't need the 7- or 8-page feature list, or the 30 (or so) page reference design document. I need register lists and theory of operation and a discussion of power modes. The chip was end-of-lifed in 2004. Many thanks, Chris

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[alsa-devel] v1.0.14 - test packages #2
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hi, I recreated driver and tools packages in /pub/testing to fix reported problems. Please, test... Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Problem with ncurses and alsamixer (cannot open terminal: vt100 or xterm)
by David Topper 04 Jun '07

04 Jun '07

I finally managed to get the alsa-1.0.4 utils to compile on an ARM based SBC. When I launch alsamixer, however, I get an error: Error opening terminal: xterm When I ssh into the board. Or: Error opening terminal: vt100 When connected via serial cable. I suspect this has something to do with the linking (or lack thereof) of alsamixer for ncurses. The ncurrses library, by the way, I cross compiled with the following command: CC=/opt/crosstool/arm-unknown-linux-gnu/gcc-3.3.4-glibc-2.3.2/bin/arm-unknown-linux-gnu-gcc ./configure --host=arm-linux --prefix=/home/topper/ts-packages/ --exec-prefix=/home/topper/ts-packages/ But for the alsa utils, I simple ran ./configure, and the script detected the ncurses library, ran through and compiled. Yet, something seems amuck because: root@ts7000:root# ldd /bin/alsamixer libasound.so.2 => /lib/libasound.so.2 (0x2aac8000) libm.so.6 => /lib/libm.so.6 (0x2ab6f000) libdl.so.2 => /lib/libdl.so.2 (0x2abe9000) libpthread.so.0 => /lib/libpthread.so.0 (0x2abf3000) libc.so.6 => /lib/libc.so.6 (0x2ac4b000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x2aaab000) Whereas on my Linux Intel laptop: awn:~$ ldd /usr/bin/alsamixer libncurses.so.5 => /lib/libncurses.so.5 (0x4002c000) libasound.so.2 => /usr/lib/libasound.so.2 (0x40069000) libm.so.6 => /lib/libm.so.6 (0x4010d000) libdl.so.2 => /lib/libdl.so.2 (0x40130000) libpthread.so.0 => /lib/libpthread.so.0 (0x40133000) libc.so.6 => /lib/libc.so.6 (0x40184000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x40000000) So I have two questions: 1. How might ensure proper linking of the ncurses library. 2. Is there another way to tell alsa to activate various cards, set volume, etc.. so I can test to see if the alsa driver is even working on this board. Thanks, Dave Topper -- Technical Director - Virginia Center for Computer Music http://www.virginia.edu/music/VCCM/

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[alsa-devel] 1.0.14 test packages
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hello all, I put 1.0.14-final test packages to ftp://ftp.alsa-project.org/pub/testing Note that these packages are not an official release. They are indended for compilation tests. Anyway, if you find a compilation (not functionality) issue, please, let me know ASAP through e-mail. I'm going to release 1.0.14 packages on next Monday. Note for alsa developers: Please, no more HG repository updates until packages are released. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] snd_hda_intel: Conexant CX20551 (Waikiki)
by Edwin Smulders 03 Jun '07

03 Jun '07

Hi, I believe there's still a problem with the HDA Intel chip Conexant CX20551 (Waikiki) not working. However, im not entirely sure the problem the problem is with alsa, because it starts working when i boot with acpi=off (a fix suggested on the ubuntu forums, but not quite useful on a laptop [plus, it breaks my ethernet]) The reason I'm mailing is, can you confirm this is something alsa can/should/will fix or are we all looking in the wrong direction and this is really an acpi problem? If you can fix this or know of something that might work, I would be happy to hear it. I've attached my lspci -nv and cat /proc/asound/card0/codec\#* just in case. I would be happy to give you any more info you might need. ~~ Edwin

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[alsa-devel] [PATCH] Add support for another X-Fi Extreme Audio (really SB Live 24bit)
by Daniel T. Chen 03 Jun '07

03 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip. Thanks, Daniel Chen

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[alsa-devel] Next steps
by chris＠idlelion.net 03 Jun '07

03 Jun '07

Hey gang, I'm trying to work on issue 305, suspend/resume for cs46xx. I'm running CentOS 5, so I'd like to work on what's current for that distribution first, and patch to latest if needed after that. I think I have the correct source package, but the only mention of cs46xx I could find in the source is a configuration file. Running make doesn't make the cs46xx module, AFAICT. Which source (package)(s) should I have? How do I track down the source file(s) I need to examine? Thanks, Chris

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Re: [alsa-devel] ASoC and a codec that can't be controlled
by Timur Tabi 02 Jun '07

02 Jun '07

Liam Girdwood wrote: >> I guess I just don't understand why the codec driver is acting like the "master" driver of >> ASOC. IMHO, the codec driver should be doing two things: > > I guess this is due to an ordering issue we had during early > development. We had to make sure that the I2C probe of the codec > succeeded before registering the card, pcms, etc. Fwiw, I do intend to > address this in future versions (I should probably add a road map to the > wiki now). In that case, I think we can solve this problem as well as the PowerPC device tree problem in one shot, because they're basically the same problem. Anyway, let me see if I get this right: 1. The first function called is eti_b1_init(). 2. eti_b1_init() calls platform_device_add() to add an soc-audio class device and register the eti_b1_snd_devdata structure. 3. The registration causes a number of things to happen, one of which is a call to soc_probe(). 4. soc_probe() sees that the snd_soc_machine_eti_b1.probe is NULL, so that part is skipped. 3. soc_probe() sees that snd_soc_machine_eti_b1.dai_link->cpu_dai->probe is also NULL, so that part is skipped. (eti_b1_snd_devdata.machine == snd_soc_machine_eti_b1) 4. soc_probe() sees that soc_codec_dev_wm8731.probe is not NULL, so it calls wm8731_probe(). 5. wm8731_probe() registers an I2C driver. 6. The I2C class driver calls wm8731_i2c_driver.attach_adapter, which is wm8731_i2c_attach(). 7. wm8731_i2c_attach() calls i2c_probe(), which in turn calls wm8731_codec_probe() 8. wm8731_codec_probe() allocates a snd_soc_codec structure. 8. wm8731_codec_probe() calls wm8731_init(). This means that if there is no I2C support, wm8731_init() will never be called. 9. wm8731_init() initializes the snd_soc_codec structure. Question: why doesn't wm8731_probe() initialize the non-I2C parts of the snd_soc_codec structure? For example, snd_soc_codec.dai, snd_soc_codec.name, and snd_soc_codec.owner? That way, the snd_soc_codec structure will be properly initialized even when there's no I2C. 10. wm8731_init() calls snd_soc_new_pcms. 11. wm8731_init() exits, wm8731_codec_probe() exits, wm8731_i2c_attach() exits, and then wm8731_probe() exits. Control returns to soc_probe(). 12. soc_probe() notices that eti_b1_snd_devdata.platform.probe is NULL, so it skips that. Question: why don't you define a function for at91_soc_platform.probe, and in that function you can call snd_soc_new_pcms()? That way, you guarantee the codec driver's I2C interface is initialized before snd_soc_new_pcms() is called, and you avoid have PCM calls in the codec driver. >> ASOC and the machine driver should then work in tandem to decide which driver will do what >> and which capabilities are *actually* supported. *Something* needs to look at the entire >> system and say to each device, "Well, yes, I know about this little feature of yours, but >> we're just not going to support that today." >> > > The machine driver pretty much already does this. It can override valid > hardware configurations and return -EINVAl if required. Do you have an example of that? Would that be eti_b1_hw_params()? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [1/3] 2.6.22-rc3: known regressions v2
by Michal Piotrowski 01 Jun '07

01 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc3. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : build failed in function `vic_sys_interrupt' References : http://lkml.org/lkml/2007/5/31/227 Submitter : Toralf Förster <toralf.foerster(a)gmx.de> Status : Unknown Subject : long freezes on thinkpad t60 References : http://lkml.org/lkml/2007/5/24/100 Submitter : Miklos Szeredi <miklos(a)szeredi.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown Networking Subject : b44: regression in 2.6.22 References : http://lkml.org/lkml/2007/5/27/108 Submitter : Maximilian Engelhardt <maxi(a)daemonizer.de> Handled-By : Michael Buesch <mb(a)bu3sch.de> Status : problem is being debugged Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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[alsa-devel] Configure problems for alsa-utils-1.0.4
by David Topper 01 Jun '07

01 Jun '07

So I've managed to compile alsa-drivers-1.0.4 for ARM. I can even load the modules. I've also successfully compiled alsa-libs-1.0.4 ... but I had to do it on the board. Cross compiling simply didn't seem to work. I've been using --prefix=/ and --exec-prefix=/ for configures. But when I do ./configure --prefix=/ --exec-prefix=/ for alsa-utils-1.0.4 it fails with: checking for libasound headers version >= 0.9.0... not present. configure: error: Sufficiently new version of libasound not found. But they're there, for sure. Perhaps it doesn't like looking in /include/alsa? Thanks, DT

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[alsa-devel] snd_soc_dai_link.init()
by Timur Tabi 01 Jun '07

01 Jun '07

Liam, Why does the init() function pointer in the snd_soc_dai_link structure take a snd_soc_codec pointer? Wouldn't it make more sense for it to take a snd_soc_dai_link pointer? That way, I could use the init() function to initialize the structure. I'm trying to write a function that will initialize the codec_dai pointer. I thought I could use snd_soc_dai_link.init(), but I don't know how to get the pointer to the snd_soc_dai_link structure from a snd_soc_codec structure. /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_dai_link { char *name; /* Codec name */ char *stream_name; /* Stream name */ /* DAI */ struct snd_soc_codec_dai *codec_dai; struct snd_soc_cpu_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_codec *codec); }; -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] Power management state flow in ALSA
by Nobin Mathew 01 Jun '07

01 Jun '07

I am having a doubt regarding ALSA power management. My understanding of APM suspend() is like this. Freeze the ALSA apps Call ALSA driver suspend () in the ALSA suspend() function it saves the current state of substream and changes the state of substream to SUSPENDED. My understanding of APM resume() is like this Call ALSA driver resume () Activate the ALSA apps In ALSA resume function it restores the saved state of substream. So my question is when ALSA app will get to know the SUSPENDED state of substream.??? I am using linux kernel 2.6.17 ALSA driver version 1.0.11rc4 ASoC patch version 0.12.4

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