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

Re: [alsa-devel] maestro3 & es1988: mic range too low
by ikke 18 Aug '08

18 Aug '08

On Mon, Aug 18, 2008 at 1:20 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 17 Aug 2008 22:07:19 +0300, > > Note that "Mic Playback Volume" controls only the volume of analog > playback. It doesn't change the recording level at all. thanks for reminding, sure. >> the reasoning for the original patch sounds reasonable. The >> mic volume is tunable, only at the very low range. I recall the laptop >> came with Win98, and there was no problem then with the sound -> not a >> HW problem. I have never got this laptop mic working with linux, so >> this is not anything that broke recently. I see there is >> check_volume_resolution function which tries to guess some of the >> register ranges, but maybe it get's it wrong with this HW. > > Maybe, maybe not. You should check rather ac97 registers in > /proc/asound/card0/codec97#0/ac97#0-0+regs. If build with > CONFIG_SND_DEBUG=y, you can write to the register directly, too. > See Documentaiton[/sound/alsa]/Procfile.txt. > I checked this file. It seems it gets the regs right for the mic (range 0x1f). The capture range is 0xf. I don't have the specs to compare it against, so I assume there is nothing to do more here. It seems mic value decreases while going up to full, where as capture increases. Another difference is maybe stereo / mono related, the capture changes both bytes whereas mic only the other one. Also one thing, should the mic level be around 0x40 - 0x5f, not 0x0 - 0x1f. Here's the diff of the registers between mixing the capture and mic from min to max: $ diff /tmp/ac97_regs_capt_{min,full} 15c15 < 0:1c = 0000 --- > 0:1c = 0f0f $ diff /tmp/ac97_regs_mic_{min,full} 8c8 < 0:0e = 005f --- > 0:0e = 0040 I'll recompile it with sound debug just to test it... Thanks, ikke

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45
by S. Aguinaga 18 Aug '08

18 Aug '08

Stan, Thank you for spending time to look at this issue. I owe you a drink, when you are in the chicago area let me know and I'll treat you to one. Ok, so I will check with the developers an see what can be done to correct this. Thank you. //Salvador 1. Re: latest Mac Pro (quad core) audio issues (stan) Message: 1 Date: Fri, 15 Aug 2008 15:14:12 -0700 From: stan <ghjeold_i_mwee(a)cox.net> Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues To: "S. Aguinaga" <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Message-ID: <48A5FFB4.10700(a)cox.net> Content-Type: text/plain; charset=ISO-8859-1; format=flowed S. Aguinaga wrote: > > Hello Stan, > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > 2. aplay does not produce sound, period! > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > sudo cat /var/log/messages | grep ALSA > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped > > Let me know if I should send something else. > > // Salvador > // Northwestern University > I think your problem is a misidentification of your sound chip. The model being used for your 889A appears to be incompatible, thus all the errors above. It appears that the macpro ALC885 is being selected before it gets to the choice for the ALC889A. Perhaps a developer can look into this. !!HDA-Intel Codec information !!--------------------------- Codec: Realtek ALC889A !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC885 Analog [ALC885 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 One shows ALC889A and the other shows ALC885. In the driver code there is a comment to the effect that the 889A is best handled as an ALC888, yet the macpro is listed under the ALC885. I am not a developer so can't go beyond this, but it seems that there is a problem. There should be a macpro under the ALC888 as well as the 885. Here are the model options for the two candidates ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 targa Targa T8, MSI-1049 T8 asus-a7j ASUS A7J asus-a7m ASUS A7M macpro MacPro support mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) ALC883/888 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O 3stack-6ch 3-jack 6-channel 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) acer-aspire Acer Aspire 9810 medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI targa-2ch-dig Targs/MSI with 2-channel laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) lenovo-101e Lenovo 101E lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D clevo-m720 Clevo M720 laptop series fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) _From the code, here is the comment indicating it should be treated as an ALC888: if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { /* Pick up systems that don't supply PCI SSID */ switch (codec->subsystem_id) { case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; break; default: /* ALC889A is handled better as ALC888-compatible */ if (codec->revision_id == 0x100103) { alc_free(codec); return patch_alc883(codec); } printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); board_config = ALC882_AUTO; } } ------------------------------ Message: 2 Date: Fri, 15 Aug 2008 16:15:45 -0700 From: "John L. Utz III" <john.utz(a)dmx.com> Subject: [alsa-devel] alsa git: which version of ac97_codec.c am i supposed to hxr? To: "ALSA Developers" <alsa-devel(a)alsa-project.org> Message-ID: <op.ufyawjke647oa9@jutz-gnto64> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes Hello All; I feel confused by the presence of 2 different copies of ac97_codec.c: jutz-gnto ~ # find . -name ac97_codec.c ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c worse yet,they are not in sync: jutz-gnto ~ # diff -w ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c 1d0 < #include "adriver.h" 38c37 < #include "../../alsa-kernel/pci/ac97/ac97_id.h" --- > #include "ac97_id.h" 40c39 < #include "../../alsa-kernel/pci/ac97/ac97_patch.c" --- > #include "ac97_patch.c" 1917d1915 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 0) 1929d1926 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 14) 1931d1927 < #endif 1953,1956d1948 < #else /* <= 2.6.0 */ < #define snd_ac97_dev_register NULL < #define snd_ac97_dev_disconnect NULL < #endif Which of these am i supposed to use? am i supposed to change both of them for some backward compatibility purpose? tnx! johnu ------------------------------ Message: 3 Date: Sat, 16 Aug 2008 06:27:02 +0000 (UTC) From: Simon Tremblay <bidou(a)videotron.qc.ca> Subject: [alsa-devel] Configure and use dmix with a new ASoC driver To: alsa-devel(a)alsa-project.org Message-ID: <loom.20080816T061550-625(a)post.gmane.org> Content-Type: text/plain; charset=us-ascii Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done? ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 18, Issue 45 ******************************************

3 2

Re: [alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by krzysztof.h1＠poczta.fm 18 Aug '08

18 Aug '08

> At Mon, 18 Aug 2008 10:50:15 +0200, > Rene Herman wrote: > > > > On 17-08-08 22:38, Krzysztof Helt wrote: > > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > > > Mute sound by setting mute bit without > > > setting volume to 0. It makes both source code > > > and binary shorter. > > > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > Grmmbbled-over-by: Rene Herman <rene.herman(a)gmail.com> > > > > This should be safe and the current function indeed generates crap code > > > but in the sense of always expecting the worst broken hardware variants > > > to be out there I must say I'd rather have seen that GCC were beter. But > > > yes, sure. > > Yeah, the slightest concern is the intention of using 0x80 there. > If this patch was already tested on some real hardwares (and I guess > it should work), I'm willing to apply it. Krzystof, was this tested > on your machines? > I have tested it only on cs4236 (inside laptop) and opti931 I have in the old PC at the moment. If you want I can retest on more hardware. The second check I noticed that for some registers the volume is defined as gain (so 0 is the least audible) and for some (CS4231_MONO_CTRL, CS4231_LEFT/RIGHT_OUTPUT) is defined as attentuation (the 0 is the most audible). In two cases, the old code set volume to the loudest and mute it with mute bit, probably the opposite what was intended. It is not a crucial patch. You may drop it. Regards, Krzysztof ---------------------------------------------------------------------- Wygraj laptopa lub iPoda! Wez udzial w igrzyskach! Kliknij >>> http://link.interia.pl/f1eda

2 1

[alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by Krzysztof Helt 18 Aug '08

18 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Mute sound by setting mute bit without setting volume to 0. It makes both source code and binary shorter. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp alsa-ref/sound/isa/wss/wss_lib.c alsa-driver/sound/isa/wss/wss_lib.c --- alsa-ref/sound/isa/wss/wss_lib.c 2008-08-14 00:05:30.000000000 +0200 +++ alsa-driver/sound/isa/wss/wss_lib.c 2008-08-17 22:33:24.620085406 +0200 @@ -574,7 +574,7 @@ static void snd_wss_calibrate_mute(struc { unsigned long flags; - mute = mute ? 1 : 0; + mute = mute ? 0x80 : 0; spin_lock_irqsave(&chip->reg_lock, flags); if (chip->calibrate_mute == mute) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -589,34 +589,34 @@ static void snd_wss_calibrate_mute(struc chip->image[CS4231_LOOPBACK]); } snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); + mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX1_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); + mute | chip->image[CS4231_AUX2_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX2_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); + mute | chip->image[CS4231_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + mute | chip->image[CS4231_RIGHT_OUTPUT]); if (!(chip->hardware & WSS_HW_AD1848_MASK)) { snd_wss_dout(chip, CS4231_LEFT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + mute | chip->image[CS4231_LEFT_LINE_IN]); snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + mute | chip->image[CS4231_RIGHT_LINE_IN]); snd_wss_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); } if (chip->hardware == WSS_HW_INTERWAVE) { snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); + mute | chip->image[CS4231_LEFT_MIC_INPUT]); snd_wss_dout(chip, CS4231_RIGHT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); + mute | chip->image[CS4231_RIGHT_MIC_INPUT]); snd_wss_dout(chip, CS4231_LINE_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); + mute | chip->image[CS4231_LINE_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_LINE_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); + mute | chip->image[CS4231_LINE_RIGHT_OUTPUT]); } chip->calibrate_mute = mute; spin_unlock_irqrestore(&chip->reg_lock, flags); ---------------------------------------------------------------------- Igrzyska na Dzikim Zachodzie! Sprawdz >> http://link.interia.pl/f1edc

3 3

Re: [alsa-devel] developing audio with ALSA
by Henry Ho 18 Aug '08

18 Aug '08

Hi After reading the cross-compilation instruction in INSTALL file, I already successfully configured the alsa-lib-2.0.17. I also successfully made the library. Adding --prefix=xxx during configuration can set the output directory to xxx. I also modify my Makefile for application, including the search path of libasound.so.2.***. Now I can successfully make it. However, when I run my application on the target, I got the following error message: ****************************************************************************************** ./pcm: error while loading shared libraries: libasound.so.2: cannot open shared object file: No such file or directory ****************************************************************************************** I have put libasound.so.2.0.0 in /tmp of the target. I also modified /etc/ld.so.conf, adding /tmp in ld.so.conf file. However, I cannot find ldconfig on the target. Is anyone can tell me how to solve it? Should I rebuild a busybox with this command? Henry ===================================================================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of RichWave is strictly prohibited; and any information in this email irrelevant to the official business of RichWave shall be deemed as neither given nor endorsed by RichWave.

3 2

[alsa-devel] maestro3 & es1988: mic range too low
by ikke 18 Aug '08

18 Aug '08

Hi, I'm having mic problems with snd_maestro3 and es1988 allegro hw. The laptop is HP OmniBook Xe3. The mic is way too silent (unusable) even with 20db boost option on. I found this seems to have been really common problem, but only one came with a fix [1] to the problem, enabling one more bit to volume control. Could someone please help me apply that fix to current alsa? I tried compiling 2.6.26.2 kernel and made the first change of the patch from mail [1] attachement [2]. No change. The second part of patch I did not find anymore from the code. There are similar parts in ac97_patch.c. Any hint what should be added there? The original patched code is somewhat outdated compared to modern one... But the reasoning for the original patch sounds reasonable. The mic volume is tunable, only at the very low range. I recall the laptop came with Win98, and there was no problem then with the sound -> not a HW problem. I have never got this laptop mic working with linux, so this is not anything that broke recently. I see there is check_volume_resolution function which tries to guess some of the register ranges, but maybe it get's it wrong with this HW. I tried with external headset, the behavior is just the same. The headset is working fine with IBM T41, so not due the mic itself. Any help would be preciated, thanks, ikke [1] http://thread.gmane.org/gmane.linux.alsa.devel/2025 [2] http://cache.gmane.org//gmane/linux/alsa/devel/2025-001.bin Other posts related, with no solutions: http://article.gmane.org/gmane.linux.alsa.user/9802 http://article.gmane.org/gmane.linux.alsa.user/15260 http://article.gmane.org/gmane.linux.alsa.user/9663 http://article.gmane.org/gmane.linux.alsa.devel/27195 HW is HP Omnibook Xe3, 00:08.0 Multimedia audio controller: ESS Technology ES1988 Allegro-1 (rev 12) Ubuntu up to date hardy, with manually compiled linux-2.6.26.2. the patch: ----------------------------------- --- ac97_codec.c Fri Jun 21 16:09:15 2002 +++ /usr/src/linux/sound/pci/ac97/ac97_codec.c Sun Jun 16 21:56:00 2002 @@ -579,7 +579,7 @@ static const snd_kcontrol_new_t snd_ac97_controls_mic[3] = { AC97_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), -AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 15, 1), +AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0) }; @@ -1137,7 +1137,7 @@ for (idx = 0; idx < 3; idx++) if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic[idx], ac97))) < 0) return err; - snd_ac97_write_cache(ac97, AC97_MIC, 0x801f); + snd_ac97_write_cache(ac97, AC97_MIC, 0x805f); /* build Line controls */ for (idx = 0; idx < 2; idx++) ----------------------------------- amixer settings: ----------------------------------- $ amixer Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 21 [68%] [-15.00dB] [on] Front Right: Playback 21 [68%] [-15.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control '3D Control - Center',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control '3D Control - Depth',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 15 [100%] [0.00dB] [on] Simple mixer control 'Aux',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 15 [100%] [22.50dB] [on] Front Right: Capture 15 [100%] [22.50dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] ----------------------------------- ---> looking at the mic range it seems to be 0-31 already, capture is only 0-15, I wonder if that's right...

2 1

[alsa-devel] Configure and use dmix with a new ASoC driver
by Simon Tremblay 18 Aug '08

18 Aug '08

Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done?

2 3

[alsa-devel] [PATCH] opl3sa2: drop snd_card pointer from the snd_opl3sa2 structure
by Krzysztof Helt 17 Aug '08

17 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove card pointer from the snd_opl3sa2 structure and break circular reference snd_card->snd_opl3sa2->snd_card. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- A very small opl3sa2 clean up. diff -urp linux-ref/sound/isa/opl3sa2.c linux-2.6/sound/isa/opl3sa2.c --- linux-ref/sound/isa/opl3sa2.c 2008-08-14 00:05:30.000000000 +0200 +++ linux-2.6/sound/isa/opl3sa2.c 2008-08-15 12:30:58.000000000 +0200 @@ -124,7 +124,6 @@ static int pnpc_registered; #define OPL3SA2_PM_D3 (OPL3SA2_PM_ADOWN|OPL3SA2_PM_PSV|OPL3SA2_PM_PDN|OPL3SA2_PM_PDX) struct snd_opl3sa2 { - struct snd_card *card; int version; /* 2 or 3 */ unsigned long port; /* control port */ struct resource *res_port; /* control port resource */ @@ -222,14 +221,13 @@ static void snd_opl3sa2_write(struct snd spin_unlock_irqrestore(&chip->reg_lock, flags); } -static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_detect(struct snd_card *card) { - struct snd_card *card; + struct snd_opl3sa2 *chip = card->private_data; unsigned long port; unsigned char tmp, tmp1; char str[2]; - card = chip->card; port = chip->port; if ((chip->res_port = request_region(port, 2, "OPL3-SA control")) == NULL) { snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); @@ -298,12 +296,14 @@ static int __devinit snd_opl3sa2_detect( static irqreturn_t snd_opl3sa2_interrupt(int irq, void *dev_id) { unsigned short status; - struct snd_opl3sa2 *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_opl3sa2 *chip; int handled = 0; - if (chip == NULL || chip->card == NULL) + if (card == NULL || card->private_data == NULL) return IRQ_NONE; + chip = card->private_data; status = snd_opl3sa2_read(chip, OPL3SA2_IRQ_STATUS); if (status & 0x20) { @@ -327,8 +327,10 @@ static irqreturn_t snd_opl3sa2_interrupt snd_opl3sa2_read(chip, OPL3SA2_MASTER_RIGHT); snd_opl3sa2_read(chip, OPL3SA2_MASTER_LEFT); if (chip->master_switch && chip->master_volume) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } } return IRQ_RETVAL(handled); @@ -512,9 +514,9 @@ static void snd_opl3sa2_master_free(stru chip->master_volume = NULL; } -static int __devinit snd_opl3sa2_mixer(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; + struct snd_opl3sa2 *chip = card->private_data; struct snd_ctl_elem_id id1, id2; struct snd_kcontrol *kctl; unsigned int idx; @@ -650,7 +652,6 @@ static struct snd_card *snd_opl3sa2_card chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; - chip->card = card; card->private_free = snd_opl3sa2_free; return card; } @@ -672,9 +673,12 @@ static int __devinit snd_opl3sa2_probe(s xdma2 = dma2[dev]; if (xdma2 < 0) chip->single_dma = 1; - if ((err = snd_opl3sa2_detect(chip)) < 0) + err = snd_opl3sa2_detect(card); + if (err < 0) return err; - if (request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, "OPL3-SA2", chip)) { + err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + "OPL3-SA2", card); + if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); return -ENODEV; } @@ -694,7 +698,8 @@ static int __devinit snd_opl3sa2_probe(s err = snd_wss_mixer(wss); if (err < 0) return err; - if ((err = snd_opl3sa2_mixer(chip)) < 0) + err = snd_opl3sa2_mixer(card); + if (err < 0) return err; err = snd_wss_timer(wss, 0, NULL); if (err < 0) ---------------------------------------------------------------------- Prosty kredyt na wszystkie potrzeby. Od 1000 zl do 120 000 zl. Bez poreczycieli i zabezpieczen >>> http://link.interia.pl/f1edf

2 2

[alsa-devel] Problem with latest alsa-info.sh patch
by Travis Place 17 Aug '08

17 Aug '08

The latest patch to allow the uploading to alsa-project.org seems to work fine, for the most part, unless you want to use the --pastebin option for the legacy upload to pastebin.ca . It seems that the uploaded data gets truncated. Not sure if this is the script, or the pastebin having problems. Ill look into it a bit more when i have time (its 5:30am here, and i need sleep). Just thought id point this out. I realise the 'default' behaviour is to use alsa-project.org for uploading, but i feel the other pastebin.ca mode should still function(I personally get people to upload there first, and then 'submit' card details to alsa-project.org/db/ afterwards. Thanks, Travis Place (wishie)

2 2

[alsa-devel] [SOUND/SOC] removed unused #include <version.h>
by Huang Weiyi 17 Aug '08

17 Aug '08

The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/wm8753.c sound/soc/codecs/wm9712.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8604809..dc7b18f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -34,7 +34,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fb7f9a..2f1c91b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h>

2 1