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[alsa-devel] Enabling in-kernel synch for M-Audio boards
by Ludovico Verducci 19 Sep '07

19 Sep '07

Hello all! I'm developing a complex multichannel audio distribution system where multiple linux boxes will stream audio data over ethernet and then should play audio at sample level resolution synchronization. The boxes clocks are synchronized over ethernet using PTP. I need to keep in synch the audio board's clocks and I can't use an external wordclock nor s/pdif. At the moment I'm experimenting with m-audio delta 1010, based on envy24 (VT1712). I know these boards can synch their clocks over PCI signalling (in windows drivers this feature support synching up to 4 boards), but ALSA drivers don't implement that feature. I'm trying to implement a similar feature (controlling board clock synch through kernel module), but it's very difficult without a good starting point! Is there anyone that could give me some help, support or infos? I would eventually appreciate also hardware reference designs or application notes about the VIA envy24 VT1712 chip. Obviously, should I ever be successful in the implementation, those features could be eventually added to ALSA drivers. Thank you all in advance. Regards, Ludovico

3 7

Re: [alsa-devel] Intel HDA / AD1986A problems
by Takashi Iwai 18 Sep '07

18 Sep '07

At Tue, 21 Aug 2007 21:45:52 -0400, Peter Skensved wrote: > > On Tue, Aug 21, 2007 at 02:34:22PM +0200, Takashi Iwai wrote: > > At Mon, 20 Aug 2007 22:27:40 -0400, > > Peter Skensved wrote: > > > > > > > > > > > > Configuration : Lenovo 3000 N100 laptop with Intel HDA / Anlog Devices AD1986A > > > soundcard running CentOS5.0 with 2.6.18-8.1.8.el5 kernel ( uses 1.0.12rc1 based > > > alsa driver ). > > > > > > The driver works with the model=laptop-eapd option except for a few things : > > > > > > 1) Plugging in an external headset will not mute the internal speakers ( has to be done > > > manually in alsa-mixer ) > > > > That's an unimplemented feature. It shouldn't be too hard to add it > > if you help debugging well (i.e. you can build and test drivers from > > HG tree without hitch). > > > > Ok - I'm willing to give it a try. Do I need development version of theh library > and the utilities as well ? Recommended, but not mandatory. > > > 2) The PC speaker ( bell ) appears to be permanently muted. > > > > Partly intended (as I dislike it ;) > > > > It is nice to have the option ... Another opinion :) > > > 3) The external microphone input does not appear to be working no matter what I do. > > > > Do you mean for recording or analog loopback? > > > > Yes - for recording. I was hoping to get it to work well enough to use skype - Note that recording and analog-loopback are different things. > > > 4) When recording from the internal microphone the level is very, very low and almost > > > drowns in static noise. > > > > There are "Mic Boost" and "Front Mic Boost" mixer volumes. > > > > Not with model=laptop-eapd - my alas mixer only shows Master, PCM, Mic, Capture, Mix, > External Amp and Internal Amp The Mic Boost control was added _after_ 1.0.14 release. (Correction in the above text: Front Mic Boost doesn't exist for AD1986A but for other codec.) So, please try HG version first. Then let's debug further. Takashi

2 3

[alsa-devel] AD1986A stop working if adjust the volume
by zhejiang 18 Sep '07

18 Sep '07

Hi, I have a ICH8M + AD1986A machine and run alsa-driver-1.0.15rc. I found that : In 6 channel mode,if adjust the front or pcm volume when playing music, all channel will be muted,but if adjust the volume alone(don't play music at the same time),every thing is ok. I use printk to display the verb in the azx_send_cmd(), when adjusting the volume in either case, there are just two verb (0x300) sent to the nid(0x1b) [correct action]. But in the first case, the dac widget amp-out vals are all affected. Some other interesting thing, If i adjust the Sur or CLFE volume when playing music, the printk show that correct verb is sent, but the codec status don't change. If I adjust them alone,every thing is ok. The card0/codec#0 before adjust volume : Codec: Analog Devices AD1986A Address: 0 Vendor Id: 0x11d41986 Subsystem Id: 0xbfd60000 Revision Id: 0x100500 No Modem Function Group found Default PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: 0x0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f* 0x20 0x1d 0x1d 0x27 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20* 0x1c 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo Connection: 2 0x0f* 0x2b Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 8 0x11* 0x22 0x00 0x21 0x10 0x07 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x8f] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x10 0x10] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x10 0x10] Pincap 0x0837: IN OUT Detect Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x0e 0x0f] Pincap 0x081737: IN OUT Detect Pin Default 0x01019011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x40: OUT Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x901301f0: [Fixed] Speaker at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x01813021: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0827: IN Detect Pin Default 0x99930123: [Fixed] Aux at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99330122: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90b70130: [Fixed] Telephony at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90f701f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x014511f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 8 0x07* 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f The card0/codec#0 after adjust "Front " volume: Codec: Analog Devices AD1986A Address: 0 Vendor Id: 0x11d41986 Subsystem Id: 0xbfd60000 Revision Id: 0x100500 No Modem Function Group found Default PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Power: 0x0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: 0x0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f* 0x20 0x1d 0x1d 0x27 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20* 0x1c 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo Connection: 2 0x0f* 0x2b Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 8 0x11* 0x22 0x00 0x21 0x10 0x07 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x8f] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x0837: IN OUT Detect Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x081737: IN OUT Detect Pin Default 0x01019011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x40: OUT Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f] Pincap 0x0810: OUT Pin Default 0x901301f0: [Fixed] Speaker at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x01813021: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0827: IN Detect Pin Default 0x99930123: [Fixed] Aux at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99330122: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90b70130: [Fixed] Telephony at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90f701f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x014511f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 8 0x07* 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f

2 1

[alsa-devel] [PATCH 1/2] ac97: YMF743 missing controls support
by Keita Maehara 18 Sep '07

18 Sep '07

From: Keita Maehara <maehara(a)debian.org> These patches enable some YMF743 controls (Tone/3D/IEC958) that won't be detected with the current version of ALSA. Signed-off-by: Keita Maehara <maehara(a)debian.org> --- The first one contains only cosmetic changes to share a few YMF753-specific symbols with YMF743. diff -r 4a3f9f77e676 include/ac97_codec.h --- a/include/ac97_codec.h Mon Sep 17 16:26:16 2007 +0200 +++ b/include/ac97_codec.h Tue Sep 18 00:27:24 2007 +0900 @@ -345,9 +345,9 @@ #define AC97_ALC650_GPIO_STATUS 0x78 #define AC97_ALC650_CLOCK 0x7a -/* specific - Yamaha YMF753 */ -#define AC97_YMF753_DIT_CTRL2 0x66 /* DIT Control 2 */ -#define AC97_YMF753_3D_MODE_SEL 0x68 /* 3D Mode Select */ +/* specific - Yamaha YMF7x3 */ +#define AC97_YMF7X3_DIT_CTRL 0x66 /* DIT Control (YMF743) / 2 (YMF753) */ +#define AC97_YMF7X3_3D_MODE_SEL 0x68 /* 3D Mode Select */ /* specific - C-Media */ #define AC97_CM9738_VENDOR_CTRL 0x5a diff -r 4a3f9f77e676 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Mon Sep 17 16:26:16 2007 +0200 +++ b/pci/ac97/ac97_patch.c Tue Sep 18 00:27:24 2007 +0900 @@ -205,8 +205,8 @@ static inline int is_shared_micin(struct /* The following snd_ac97_ymf753_... items added by David Shust (dshust(a)shustring.com) */ -/* It is possible to indicate to the Yamaha YMF753 the type of speakers being used. */ -static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +/* It is possible to indicate to the Yamaha YMF7x3 the type of speakers being used. */ +static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[3] = { "Standard", "Small", "Smaller" @@ -221,12 +221,12 @@ static int snd_ac97_ymf753_info_speaker( return 0; } -static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_3D_MODE_SEL]; + val = ac97->regs[AC97_YMF7X3_3D_MODE_SEL]; val = (val >> 10) & 3; if (val > 0) /* 0 = invalid */ val--; @@ -234,7 +234,7 @@ static int snd_ac97_ymf753_get_speaker(s return 0; } -static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -242,20 +242,20 @@ static int snd_ac97_ymf753_put_speaker(s if (ucontrol->value.enumerated.item[0] > 2) return -EINVAL; val = (ucontrol->value.enumerated.item[0] + 1) << 10; - return snd_ac97_update(ac97, AC97_YMF753_3D_MODE_SEL, val); -} - -static const struct snd_kcontrol_new snd_ac97_ymf753_controls_speaker = + return snd_ac97_update(ac97, AC97_YMF7X3_3D_MODE_SEL, val); +} + +static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "3D Control - Speaker", - .info = snd_ac97_ymf753_info_speaker, - .get = snd_ac97_ymf753_get_speaker, - .put = snd_ac97_ymf753_put_speaker, -}; - -/* It is possible to indicate to the Yamaha YMF753 the source to direct to the S/PDIF output. */ -static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) + .info = snd_ac97_ymf7x3_info_speaker, + .get = snd_ac97_ymf7x3_get_speaker, + .put = snd_ac97_ymf7x3_put_speaker, +}; + +/* It is possible to indicate to the Yamaha YMF7x3 the source to direct to the S/PDIF output. */ +static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[2] = { "AC-Link", "A/D Converter" }; @@ -268,17 +268,17 @@ static int snd_ac97_ymf753_spdif_source_ return 0; } -static int snd_ac97_ymf753_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val >> 1) & 1; return 0; } -static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -286,7 +286,7 @@ static int snd_ac97_ymf753_spdif_source_ if (ucontrol->value.enumerated.item[0] > 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << 1; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0002, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val); } /* The AC'97 spec states that the S/PDIF signal is to be output at pin 48. @@ -311,7 +311,7 @@ static int snd_ac97_ymf753_spdif_output_ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val & 0x0008) ? 2 : (val & 0x0020) ? 1 : 0; return 0; } @@ -325,7 +325,7 @@ static int snd_ac97_ymf753_spdif_output_ return -EINVAL; val = (ucontrol->value.enumerated.item[0] == 2) ? 0x0008 : (ucontrol->value.enumerated.item[0] == 1) ? 0x0020 : 0; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0028, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0028, val); /* The following can be used to direct S/PDIF output to pin 47 (EAPD). snd_ac97_write_cache(ac97, 0x62, snd_ac97_read(ac97, 0x62) | 0x0008); */ } @@ -334,9 +334,9 @@ static const struct snd_kcontrol_new snd { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = snd_ac97_ymf753_spdif_source_info, - .get = snd_ac97_ymf753_spdif_source_get, - .put = snd_ac97_ymf753_spdif_source_put, + .info = snd_ac97_ymf7x3_spdif_source_info, + .get = snd_ac97_ymf7x3_spdif_source_get, + .put = snd_ac97_ymf7x3_spdif_source_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -345,10 +345,10 @@ static const struct snd_kcontrol_new snd .get = snd_ac97_ymf753_spdif_output_pin_get, .put = snd_ac97_ymf753_spdif_output_pin_put, }, - AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF753_DIT_CTRL2, 2, 1, 1) -}; - -static int patch_yamaha_ymf753_3d(struct snd_ac97 * ac97) + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF7X3_DIT_CTRL, 2, 1, 1) +}; + +static int patch_yamaha_ymf7x3_3d(struct snd_ac97 * ac97) { struct snd_kcontrol *kctl; int err; @@ -358,9 +358,9 @@ static int patch_yamaha_ymf753_3d(struct strcpy(kctl->id.name, "3D Control - Wide"); kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf753_controls_speaker, ac97))) < 0) + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, ac97))) < 0) return err; - snd_ac97_write_cache(ac97, AC97_YMF753_3D_MODE_SEL, 0x0c00); + snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); return 0; } @@ -374,7 +374,7 @@ static int patch_yamaha_ymf753_post_spdi } static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { - .build_3d = patch_yamaha_ymf753_3d, + .build_3d = patch_yamaha_ymf7x3_3d, .build_post_spdif = patch_yamaha_ymf753_post_spdif };

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[alsa-devel] alsa-tools -> usx2yloader
by Elimar Riesebieter 18 Sep '07

18 Sep '07

Hi all, according to Debian #436502 I am asking herefore if there are plans to port asx2yloader from the old hotplug to udev? Debian lenny/sid relies on udev only, though. Thanks in advance Elimar -- It's a good thing we don't get all the government we pay for.

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[alsa-devel] snd_mixart_send_msg / snd_mixart_send_msg_wait_notif
by Rene Herman 18 Sep '07

18 Sep '07

Hi. While looking through ALSA for schedule_timeout() calls, I ran into: int snd_mixart_send_msg(...) { [ ... ] set_current_state(TASK_UNINTERRUPTIBLE); add_wait_queue(&mgr->msg_sleep, &wait); spin_unlock_irq(&mgr->msg_lock); timeout = schedule_timeout(MSG_TIMEOUT_JIFFIES); remove_wait_queue(&mgr->msg_sleep, &wait); if (! timeout) { /* error - no ack */ mutex_unlock(&mgr->msg_mutex); snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame); return -EIO; } [ ... ] } and the same in snd_mixart_send_msg_wait_notif(). I believe there to be something wrong with this code. A schedule_timeout() in TASK_UNINTERRUPTIBLE is always going to return 0. Didn't you intend to use a wait_event_timeout() or something like that? Puzzled, since you err out on !timeout, and it seems every run through this would end up there. Rene.

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[alsa-devel] [PATCH] cs4231: remove two auto-calibrations during chip initialization
by Krzysztof Helt 18 Sep '07

18 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The initialization function did auto-calibration after each register setting. This patch merges more register changes before auto-calibration is done. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is in the init() function so it is called just once. The patch is about cutting fat from cs4231 drivers as this change does not affect how they operate in any major way. diff -urp alsa-driver-1.0.15rc2/sound/isa/cs423x/cs4231_lib.c linux-2.6.23/sound/isa/cs423x/cs4231_lib.c --- alsa-driver-1.0.15rc2/sound/isa/cs423x/cs4231_lib.c 2007-09-12 13:33:51.000000000 +0200 +++ linux-2.6.23/sound/isa/cs423x/cs4231_lib.c 2007-09-18 08:37:08.000000000 +0200 @@ -652,15 +652,7 @@ static void snd_cs4231_init(struct snd_c CS4231_CALIB_MODE); chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("init: (2)\n"); -#endif - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_down(chip); @@ -676,15 +668,6 @@ static void snd_cs4231_init(struct snd_c snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("init: (4)\n"); -#endif - - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_down(chip); diff -urp linux-2.6.23.ref/sound/sparc/cs4231.c linux-2.6.23/sound/sparc/cs4231.c --- linux-2.6.23.ref/sound/sparc/cs4231.c 2007-09-18 09:26:12.000000000 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-18 09:26:54.000000000 +0200 @@ -733,15 +733,6 @@ static void __init snd_cs4231_init(struc CS4231_CALIB_MODE); chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (2)\n"); -#endif - - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->lock, flags); snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); @@ -757,15 +748,6 @@ static void __init snd_cs4231_init(struc snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT]); - spin_unlock_irqrestore(&chip->lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (4)\n"); -#endif - - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->lock, flags); snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip);

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[alsa-devel] [PATCH] ad1848: remove redundant autocalibration
by Krzysztof Helt 18 Sep '07

18 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch removes redundant auto-calibration (mce up/down sequence) from snd_ad1848_open(). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The docs says that changing the Interface Configuration register and the Format/Clock register requires the MCE bit set. However, the docs do not say that additional auto-calibration is required after setting one register and before setting the other. Now, both registers are set at once. Tested on SC-6000. This should reduce opening latency by one auto-calibration period. diff -urp linux-2.6.22.old/sound/isa/ad1848/ad1848_lib.c linux-2.6.23/sound/isa/ad1848/ad1848_lib.c --- linux-2.6.22.old/sound/isa/ad1848/ad1848_lib.c 2007-09-12 13:33:51.000000000 +0200 +++ linux-2.6.23/sound/isa/ad1848/ad1848_lib.c 2007-09-18 00:36:33.000000000 +0200 @@ -408,17 +408,8 @@ static int snd_ad1848_open(struct snd_ad AD1848_CALIB_MODE); chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("open: (2)\n"); -#endif snd_ad1848_set_data_format(chip, NULL); - - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); snd_ad1848_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_ad1848_mce_down(chip);

1 0

[alsa-devel] E-MU 0404/0202 USB implementation
by James Courtier-Dutton 18 Sep '07

18 Sep '07

Hi, I am looking to implement support for the E-Mu USB cards. The card is basically following a mix of USB 1.0 and USB 2.0 descriptors. The bInterfaceClass field of the standard interface descriptors for the Audio Control, Audio Streaming, and MIDI Streaming interfaces are set to 0xFF (vendor-specific) instead of 0x01 (USB Audio). Can I put in a quirk into the snd-usb-audio driver to handle this? The card also has customised controls. Can one add those to snd-usb-audio, or would it be better to add those to a totally new driver? Kind Regards James

2 1

[alsa-devel] Current ad1848/cs4231 mce_down code
by Rene Herman 18 Sep '07

18 Sep '07

By the way, while looking at the current mce_down code: http://hg.alsa-project.org/alsa-kernel/file/0028e39ead78/isa/ad1848/ad1848_… http://hg.alsa-project.org/alsa-kernel/file/0028e39ead78/isa/cs423x/cs4231_… we have the ad1848 loops doing: end_time = jiffies + msecs_to_jiffies(250); while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { spin_unlock_irqrestore(&chip->reg_lock, flags); (---) if (time_after(jiffies, end_time)) { snd_printk(KERN_ERR " [ timeout ] ") return; } msleep(1); spin_lock_irqsave(&chip->reg_lock, flags); } At (---) we are no longer atomic, so imagine being pre-empted there. When we get back, we may/will well be past end_time and although by now calibration has long finished, we bail out believing we've timed out. Should it rather be this (probably with a "goto eror" type of setup, but as far as the logic is concerned)? end_time = jiffies + msecs_to_jiffies(250); while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { if (time_after(jiffies, end_time)) { spin_unlock_irqrestore(&chip->reg_lock, flags); snd_printk(KERN_ERR " [ timeout ] ") return; } spin_unlock_irqrestore(&chip->reg_lock, flags); msleep(1); spin_lock_irqsave(&chip->reg_lock, flags); } Moreover, we aren't under lock at all in the cs4231 version (which also means that msleep(1) thing is in fact fine there). Can't we do that for ad1848 as well? (but as said, minimal fix for now is the msleep(1) under lock fix in ad1848) Rene.

2 2