March 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Acer Ferrari 5000
by Russ Dill 24 Mar '09

24 Mar '09

GIven the recent changes in gnome land (pulseaudio), I'd like to be able to get alsa support for my hardware fixed up. My biggest annoyance right now is that when I plug headphones in, it doesn't mute the speakers. Another issue I have is that I'd like to have support for audio out on HDMI (which is supported by Vista). On with the show. I'm currently running 2.6.28-9-generic (Ubuntu 9.04). All my info is below, what can I do to get everything exposed and working? 00:14.2 Audio device: ATI Technologies Inc IXP SB4x0 High Definition Audio Controller (rev 01) Subsystem: Acer Incorporated [ALI] Device 010a Flags: bus master, slow devsel, latency 64, IRQ 16 Memory at b0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel HDA Intel 0000:00:14.2: PCI INT A disabled HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 hda-intel: Invalid position buffer, using LPIB read method instead. hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x101f000a hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x101f000a (methinks these timeouts are no good) devices: 2: : timer 3: [ 0- 4]: digital audio capture 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control pcm0c/info: card: 0 device: 0 subdevice: 0 stream: CAPTURE id: ALC883 Analog name: ALC883 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 pcm0p/info: card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: ALC883 Analog name: ALC883 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 pcm4c/info: card: 0 device: 4 subdevice: 0 stream: CAPTURE id: ALC883 Analog name: ALC883 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 codec#0: Codec: Conexant ID 2bfa Address: 0 Vendor Id: 0x14f12bfa Subsystem Id: 0x1025010a Revision Id: 0x90000 Modem Function Group: 0x2 codec#1: Codec: Realtek ALC883 Address: 1 Vendor Id: 0x10ec0883 Subsystem Id: 0x10ec0883 Revision Id: 0x100002 No Modem Function Group found Default PCM: N/A Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x98 0x98] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x15 0x15] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x15 0x15] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01014030: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01011031: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01016032: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x3, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01012033: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x3, Sequence = 0x3 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19850: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19c80: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x8, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x01813050: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02214c40: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x99331150: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Black DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x00000100: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x01441170: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x01c46160: [Jack] SPDIF In at Ext Rear Conn = RCA, Color = Orange DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

2 22

[alsa-devel] A small annoyance with alsa-driver
by "Rafał Mużyło" 23 Mar '09

23 Mar '09

As your bugtrack seem unresponsive, I'm asking here: could you look at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4274 and either correct the docs or make stuff work like it did before ?

1 0

[alsa-devel] trouble using dmix with tts systems (accessibility )
by Halim Sahin 23 Mar '09

23 Mar '09

hello, I have asked this question to alsa user list but got no useful answer. I am a blind linux user and need asistive Technology to use my pc. The problem I have experienced with alsa is that it is sometimes too slow to stop previous output before starting a new one. The result of this ends in several speech outputs at a time. The problem only occurs if the dmix plugin is used. Is there a way to modify the buffer to avoid these delays??? I have tried to customize my buffer_size vallue in ~/.asoundrc but I wasn't able to to create a working setup with dmix. All help appreciated ;-) Regards halim

2 2

[alsa-devel] [PATCH] alsa: Intel HDA output and align Ids.
by Pascal de Bruijn | Unilogic Networks B.V. 23 Mar '09

23 Mar '09

This patch does two things: Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros Before: Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x100100 After: Function Id: 0x1 Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x0100100 As report on the Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn <pascal(a)unilogicnetworks.net> diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.c linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.c --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.c 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.c 2009-03-23 10:37:30.000000000 +0100 @@ -647,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, + AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.h linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.h --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.h 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.h 2009-03-20 16:10:07.000000000 +0100 @@ -739,6 +739,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_proc.c linux-2.6.29-rc8-func/sound/pci/hda/hda_proc.c --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_proc.c 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_proc.c 2009-03-20 16:10:07.000000000 +0100 @@ -467,8 +467,9 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "Codec: %s\n", codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); if (codec->mfg)

2 1

[alsa-devel] How to find sampling rate supported
by reetesh chandran 23 Mar '09

23 Mar '09

Hi all, I am new to the group and to Alsa. For our project, we have to find the max sampling rate and other pcm hardware parameters that the connected USB headset supports. I looked at the alsa library, but could not find an API which returned this information. I found snd_pcm_hw_params_test_* function to test a specific value is supported, but I was wondering if there was a way to query the device to get the values. Thanks in advance, Reetesh

2 1

Re: [alsa-devel] Bugs on aspire one A150
by Andreas Mohr 23 Mar '09

23 Mar '09

Hi, On Mon, Mar 16, 2009 at 05:19:38PM +0100, Takashi Iwai wrote: > At Mon, 16 Mar 2009 17:06:35 +0100, > 私 wrote: > > > > > > What are "sliders"? > > > > > > Umm, volume level controls. > > > > Yes but there are many of such :) > > > > More exactly, from the driver perspective, there are no volume > > controls but only there are control elements with integer values. > > Do you mean "Capture Volume" control or which one? Hmm, ok, this needs to be more precise: In gamix (codec "HDA Intel : Realtek ALC268"), the Capture Volume control. > > And, is the behavior consistent regardless of the value high, i.e. > > the key is only whether the values for both channels are identical? > > BTW, what if you record with the following definition? > Put the below to ~/.asoundrc > > pcm.imix { > type plug > slave.pcm "hw" > ttable.0.0 0.5 > ttable.0.1 -0.5 > } > > and record like > > % aplay -Dimix -c1 foo.wav Does NOT exhibit the "equal sliders == no sound" bug (apart from this sliders are acting normally, i.e. slider low == no sound), despite being a "plug" type definition (this is what you wanted to discern, right? ;). Thanks! Andreas

3 23

[alsa-devel] MSI GX620 support
by okias 22 Mar '09

22 Mar '09

MSI GX620X-061CZ, but probably is this audio chip with this configuration on all GX620 Connectors: Green, standart jack 3.5 headphones Blue, line-in Pink, mic Grey, side DOLBY HOME THEATER KO 127 3stack-dig 3-jack with SPDIF I/O *OK 128 6stack-dig 6-jack digital with SPDIF I/O // work SIDE+jack, but SIDE can't be configured, not deactivating integrated repro *OK 129 3stack-6ch 3-jack 6-channel // headphone power off integrated repro, 6ch working, no 8ch OK 130 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O //same 3stacj-6ch KO 131 6stack-dig-demo 6-jack digital for Intel demo board OK 132 acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)//jack+side, but SIDE can't be configured, headphone power off integr., notworking 6ch KO 133 acer-aspire Acer Aspire 9810 *OK 134 acer-aspire-4930g Acer Aspire 4930G // Front switch shutting down both repro (external in jack, integrated), 6channels + SIDE (should be 8ch), and if is plugged something into jack integrated repro still playing KO 135 medion Medion Laptops KO 136 medion-md2 Medion MD2 !! 137 targa-dig Targa/MSI // If is plugged jack, then integrated repro power down! but missing settings to keep integrated repro playing and 8ch isn't present(side not working). KO 138 targa-2ch-dig Targs/MSI with 2-channel KO 139 laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) // SIDE working, but 6ch conf not, only side + repro output KO 140 lenovo-101e Lenovo 101E // nothing new... less settings, no side... no 6ch KO 141 lenovo-nb0763 Lenovo NB0763 KO 142 lenovo-ms7195-dig Lenovo MS7195 KO 143 lenovo-sky Lenovo Sky KO 144 haier-w66 Haier W66 KO 145 3stack-hp HP machines with 3stack (Lucknow, Samba boards) // 4. channel not working, etc. KO 146 6stack-dell Dell machines with 6stack (Inspiron 530) // if is activated 8ch conf, side is activated, but 4 and 6th channel not working with this conf KO 147 mitac Mitac 8252D KO 148 clevo-m720 Clevo M720 laptop series KO 149 fujitsu-pi2515 Fujitsu AMILO Pi2515 *OK 150 fujitsu-xa3530 Fujitsu AMILO XA3530 // 6ch mode working, but SIDE working even if is set only 6ch KO 151 3stack-6ch-intel Intel DG33* boards OK 152 auto auto-config reading BIOS (default) // SIDE+JACK, BOTH MIC OK, can't be configured to 4,6 or 8ch, if is pluged into JACK repro then integrated playing... OK means, it's in usable state, KO mean something important doesn't work, or nothing new in compare to before tested configurations. *OK means very good configurations 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Micro-Star International Co., Ltd. Device 6510 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f8ef8000 (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] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel If you need any addional information, testing patchs (gentoo), please contact me :-) okias -- Jabber/XMPP: okias(a)isgeek.info Webové stránky: http://www.jabber.xf.cz

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[alsa-devel] [PATCH] alsa: Intel HDA output and align Ids.
by Pascal de Bruijn | Unilogic Networks B.V. 20 Mar '09

20 Mar '09

Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros As submitted to Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn <pascal(a)unilogicnetworks.net> diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_codec.c linux-2.6.28.8-func/sound/pci/hda/hda_codec.c --- linux-2.6.28.8-orig/sound/pci/hda/hda_codec.c 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_codec.c 2009-03-17 10:39:45.000000000 +0100 @@ -519,9 +519,8 @@ static void __devinit setup_fg_nodes(str total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_codec.h linux-2.6.28.8-func/sound/pci/hda/hda_codec.h --- linux-2.6.28.8-orig/sound/pci/hda/hda_codec.h 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_codec.h 2009-03-17 10:39:45.000000000 +0100 @@ -693,6 +693,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_proc.c linux-2.6.28.8-func/sound/pci/hda/hda_proc.c --- linux-2.6.28.8-orig/sound/pci/hda/hda_proc.c 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_proc.c 2009-03-17 10:40:08.000000000 +0100 @@ -520,9 +520,10 @@ static void print_codec_info(struct snd_ snd_hda_get_codec_name(codec, buf, sizeof(buf)); snd_iprintf(buffer, "Codec: %s\n", buf); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); - snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); + snd_iprintf(buffer, "Revision Id: 0x%08x\n", codec->revision_id); if (codec->mfg) snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg);

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Re: [alsa-devel] [PATCH 6/9] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Medisetty, Naresh 19 Mar '09

19 Mar '09

> On Mon, Mar 16, 2009 at 08:05:25AM -0400, Naresh Medisetty wrote: > > > + CLK("soc-audio.0", NULL, &mcasp0_clk), > > + CLK("soc-audio.1", NULL, &mcasp1_clk), > > Assuming this is a shorthand for defining two platform devices for two > ASoC cards I'd suggest that you instead move towards registering clocks > for the DAIs and probing them as platform devices - take a look at how > the PXA AC97 driver now does this for an example. The soc-audio device > will hopefully get removed at some point and you can't guarantee that > the user won't want to put the two DAIs into a single sound card. Yes, This is a shorthand for defining two platform devices for two ASoC cards. I will re-work on this and submit a patch

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[alsa-devel] [PATCH 1/9] ARM: DaVinci: ASoC: Adds ASoC driver support for TI DM646X EVM platform
by Naresh Medisetty 19 Mar '09

19 Mar '09

Adds ASoC driver support for TI DM646X EVM platform Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. sound/soc/davinci/davinci-evm.c | 131 ++++++++++++++++++++++++++++++++++++--- 1 files changed, 121 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 95c44bf..9e832c9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -25,10 +25,15 @@ #include <mach/asp.h> #include <mach/edma.h> #include <mach/mux.h> +#include <mach/cpu.h> + #include "../codecs/tlv320aic3x.h" +#include "../codecs/codec_stubs.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-i2s-mcasp.h" + #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -43,14 +48,14 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (cpu_is_davinci_dm355() || cpu_is_davinci_dm646x()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by * board-dm644x-evm.c using GPIOs from U18. There are six * options; here we "know" we use a 48 KHz sample rate. */ - else if (machine_is_davinci_evm()) + else if (cpu_is_davinci_dm644x()) sysclk = 12288000; else @@ -144,6 +149,25 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = davinci_iis_mcasp_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "MCASP SPDIF", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[1], + .codec_dai = dit_stub_dai, + .ops = &evm_ops, + }, +}; + + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +176,27 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = 2, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -165,6 +204,14 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, +}; + + /* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ static struct resource evm_snd_resources[] = { { @@ -174,11 +221,64 @@ static struct resource evm_snd_resources[] = { }, }; +static struct resource dm6467_evm_snd_resources[] = { + { + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, +}; + + +static u8 dm6467_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm6467_dit_serializer_direction[] = { + TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + + static struct evm_snd_platform_data evm_snd_data = { .tx_dma_ch = DAVINCI_DMA_ASP0_TX, .rx_dma_ch = DAVINCI_DMA_ASP0_RX, }; +static struct evm_snd_platform_data dm6467_evm_snd_data[] = { + { + .clk_name = "McASPCLK0", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .rx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = 4, + .tdm_slots = 2, + .serial_dir = dm6467_iis_serializer_direction, + .eventq_no = EVENTQ_0, + .codec_fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + { + .clk_name = "McASPCLK1", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .rx_dma_ch = -1, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = 4, + .tdm_slots = 32, + .serial_dir = dm6467_dit_serializer_direction, + .eventq_no = EVENTQ_0, + .codec_fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, +}; + + /* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ static struct resource dm335evm_snd_resources[] = { { @@ -199,23 +299,33 @@ static int __init evm_init(void) { struct resource *resources; struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; + int res_size; - if (machine_is_davinci_evm()) { + if (cpu_is_davinci_dm644x()) { davinci_cfg_reg(DM644X_MCBSP); - + evm_snd_dev_data = &evm_snd_devdata; resources = evm_snd_resources; data = &evm_snd_data; + res_size = ARRAY_SIZE(evm_snd_resources); index = 0; - } else if (machine_is_davinci_dm355_evm()) { + } else if (cpu_is_davinci_dm355()) { /* we don't use ASP1 IRQs, or we'd need to mux them ... */ davinci_cfg_reg(DM355_EVT8_ASP1_TX); davinci_cfg_reg(DM355_EVT9_ASP1_RX); - + evm_snd_dev_data = &evm_snd_devdata; resources = dm335evm_snd_resources; + res_size = ARRAY_SIZE(dm335evm_snd_resources); data = &dm335evm_snd_data; index = 1; + } else if (cpu_is_davinci_dm646x()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + resources = dm6467_evm_snd_resources; + res_size = ARRAY_SIZE(dm6467_evm_snd_resources); + data = dm6467_evm_snd_data; + index = 0; } else return -EINVAL; @@ -223,11 +333,12 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(data)); + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = data; - ret = platform_device_add_resources(evm_snd_device, resources, 1); + ret = platform_device_add_resources(evm_snd_device, + resources, res_size); if (ret) { platform_device_put(evm_snd_device); return ret; -- 1.5.6

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