June 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] ASoC: adau1701: signedness bug in adau1701_write()
by Dan Carpenter 20 Jun '11

20 Jun '11

"ret" is supposed to be signed here. The current code will only return -EIO on error, instead of a more appropriate error code such as -EAGAIN etc. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 6c01bb6..2758d5f 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -140,9 +140,10 @@ static unsigned int adau1701_register_size(struct snd_soc_codec *codec, static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - unsigned int i, ret; + unsigned int i; unsigned int size; uint8_t buf[4]; + int ret; size = adau1701_register_size(codec, reg); if (size == 0)

4 3

[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for an HP device
by David Henningsson 20 Jun '11

20 Jun '11

The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. Relevant codec info is in http://bugs.launchpad.net/bugs/761022 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH 1/3] ALSA: VIA HDA: Modify initial verbs list for VT1718S.
by lydiawang 20 Jun '11

20 Jun '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Modify initial verbs list for VT1718S. Remove some invalid initial verbs and correct some wrong initial verbs for VT1718S codec. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2011-06-20 13:25:07.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2011-06-20 13:31:19.000000000 +0800 @@ -4286,9 +4286,6 @@ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - - /* Setup default input of Front HP to MW9 */ - {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, /* PW9 PW10 Output enable */ {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, @@ -4297,10 +4294,10 @@ /* Enable Boost Volume backdoor */ {0x1, 0xf88, 0x8}, /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -4310,8 +4307,6 @@ /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Unmute MW4's index 0 */ - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } };

2 1

[alsa-devel] Via VT2020: issues with kernel 2.6.38.{2, 3} (alsa 1.0.23) - working with 2.6.33.2 (alsa 1.0.21)
by alex dot baldacchino dot alsasub at gmail dot com 20 Jun '11

20 Jun '11

Hello, First of all let me apologize if I'm not fulfilling any requirements for this list, and for any errors in my English. I've sent a mail with same subject and content to this list as a non-subscriber on April, 26, but perhaps that's gone lost, or perhaps is still under moderation or didn't pass it, so I'm resending this from a different address as a subscriber. So, please, ignore my previous mail if it came out from moderation. As per subject, I'm experiencing problems with my motherboard built-in audio device (an Asrock with ATI SB8xx), embedding a Via VT2020 codec (I haven't tested the ATI HDMI part in same conditions). The card is recognized 'about' correctly (later in this mail something about dmesg and other issues), and stuff in /proc/asound seems correct, also I have no apparent problem with /dev special files, but I get no sound after having installed kernel 2.6.38.2 and updated to version 2.6.38.3 (both compiled from source); no mixer seems to work (later specific errors), but everything still works with older kernel 2.6.33.2 I've kept in my system (built with the same build environment). Since I'm using the very same userspace environment in both cases, I deem it to be a driver(s) related issue more likely; though, I've updated my alsa-lib and -utils to latest versions (were such when I downloaded sources - alsa-lib 1.0.24.1 & alsa-utils 1.0.24.2) and still get the very same results. But let me go into more details. Here is an excerpt from dmesg called with kernel 2.6.33.2 (sound works, but I'm just using headphone jack, so I can't tell about anything related with later issue with smart5.1 and hda I've seen in list archives): [...] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19 HDA Intel 0000:01:05.1: irq 28 for MSI/MSI-X HDA Intel 0000:01:05.1: setting latency timer to 64 [...] ohci1394 0000:04:00.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 ohci1394 0000:04:00.0: setting latency timer to 64 [...] ohci1394: fw-host0: OHCI-1394 1.1 (PCI): IRQ=[16] MMIO=[fe9ff800-fe9fffff] Max Packet=[2048] IR/IT contexts=[4/8] ieee1394: Host added: ID:BUS[0-00:1023] GUID[008f130043910900] [...] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0001 [...] Notice last line is an harmless warning (sound works) and is there because msi seems to be enabled by default; by adding "options snd-hda-intel enable_msi=0" to /etc/modprobe.d/alsa-base.conf both the warning and any reference to MSI/MSI-X disappear; same effect with pci=nomsi on command line (this time affecting other drivers, such as r8169). VT2020 is part of device 0000:00:14.2 sharing IRQ 16 with ohci1394; device 0000:01:05.1 is ATI HDMI, while device 0000:05.0 should be embedded radeon vga (bridged together according with lspci -t with 0000:00:01.0 on top); for each bridge I can find with lspci -t corresponding msi_bus file in sys filesystem (in /sys/bus/pci/devices/*) is always 1, regardless of any boot command (this with kernel 2.6.33.2). Now an excerpt from dmesg in kernel 2.6.38.3: [...] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 hda-codec: no NID for mapping control Independent HP:0:0 HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19 HDA Intel 0000:01:05.1: setting latency timer to 64 [...] r8169 0000:03:00.0: irq 43 for MSI/MSI-X [...] firewire_ohci 0000:04:00.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 firewire_ohci 0000:04:00.0: setting latency timer to 64 [...] firewire_ohci: Added fw-ohci device 0000:04:00.0, OHCI v1.10, 4 IR + 8 IT contexts, quirks 0x11 firewire_core: created device fw0: GUID 008f130043910900, S400 [...] Now firewire_ohci replaces ohci1394, again sharing IRQ 16 with hda-intel for vt2020. Line about r8169 is just for reference, since it gets a different irq for msi now. Similar behaviour for MSI: enabled/disabled with pci=nomsi (seems to be enabled by default without pci=msi) and msi_bus == 1 for bridges regardless of boot options, *but for bridge on top of ati hdmi device*: now msi seems to be disabled regardless of both 'enable_msi' option and pci=* boot command and corresponding bridge (device at bus 0000:00:01.0) holds a value of 0 for msi_bus... Notice there are no more warnings about response timeouts, as it happened in older kernel after disabling msi (both globally and for alsa through alsa config options). Notice also line "hda-codec: no NID for mapping control Independent HP:0:0". That's because function 'snd_hda_get_connections' (patch_via.c) called by 'via_hp_build' returns 3 ( test for 'nums <= 1' fails and the function doesn't return), but function 'side_mute_channel', called to init knew->subdevice when 'registering' via_hp_mixer[1], returns 0 (default value) and so condition 'if(nid > 0)', in function 'snd_hda_add_nid' (hda_codec.c), is not met and the error is triggered (by the way, param nid is of type hda_nid_t which in turn is an unsigned integer (u16), so, should that be 'if(nid != 0)' instead? just to avoid the risk of giving the impression that negative values are possible but mistaken (hda_nid_t is defined elsewhere). Also: param @nid, in function comment/documentation is defined as optional, but it doesn't seem to be). Incidentally, in /proc/asound/card0/codec#0 node 0x35 looks much like node 0x34 (with both older and newer kernel; posting excerpt for kernel 2.6.38.3): Node 0x34 [Audio Selector] wcaps 0x300501: Stereo Control: name="Independent HP", index=0, device=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 3 0x08 0x0b 0x0c* Node 0x35 [Audio Selector] wcaps 0x300501: Stereo Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 3 0x08 0x0b* 0x0c (notice connections and wcaps similarity, if relevant at all) Since I'm quite crazy, I've tried to modify 'side_mute_channel' to have it returning '0x35' for codec type 'VT1718S', but to no avail; I also tried to bypass 'snd_hda_add_nid', both avoiding 'via_hp_build' to clone via_hp_mixer[<something>] and/or modifying 'via_build_controls' and 'snd_hda_add_new_ctls' (hda_control.c) so that "Independent HP" ctls were handled by the latter (achieving this was just a matter of a pair of strcmp's to choose skipping the rest of an iteration or not), as I deem it was done in alsa 1.0.21 (embedded in kernel 2.6.33.2), but it didn't work (the only result I got this ways was to avoid that error message about nids in system logs). Additional infos from userspace applications: mplayer works fine with kernel 2.6.33.2 (again, only using one exit, I haven't tested all jacks combinations); called from cmdline (under X - but that's the same shutting X down) with default options ('mplayer <path/to/file>') it gives the following output (relevant excerpt, as I deem it, after codec choice, can provide more if needed): socket(): Address family not supported by protocol AO: [pulse] Init failed: Connection refused Failed to initialize audio driver 'pulse' AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Starting playback... (notice I'm not using alsa-tools, such as pulse-audio; with the above invocation sound/video works) running mplayer with option '-ao sdl' (using sdl audio, same file) I get the following output: [AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le AO: [sdl] 44100Hz 2ch s16le (2 bytes per sample) Starting playback... Again, sound works. In both cases, I can notice a change in /proc/asound/card0/codec#0 as follows: In nodes 0x08, 0x09, 0x0a, 0x0e (all [Audio Output] nodes, first 3 with 'wcaps 0x41d: Stereo Amp-Out', last with 'wcaps 0x611: Stereo Digital'), the following line: Converter: stream=0, channel=0 changes into: Converter: stream=5, channel=0 With kernel 2.6.38.x I get the following: running mplayer with no option, I get no sound and following output: socket(): Address family not supported by protocol AO: [pulse] Init failed: Connection refused Failed to initialize audio driver 'pulse' [AO_ALSA] alsa-lib: confmisc.c:768:(parse_card) cannot find card '0' [AO_ALSA] alsa-lib: conf.c:4184:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device [AO_ALSA] alsa-lib: confmisc.c:392:(snd_func_concat) error evaluating strings [AO_ALSA] alsa-lib: conf.c:4184:(_snd_config_evaluate) function snd_func_concat returned error: No such device [AO_ALSA] alsa-lib: confmisc.c:1251:(snd_func_refer) error evaluating name [AO_ALSA] alsa-lib: conf.c:4184:(_snd_config_evaluate) function snd_func_refer returned error: No such device [AO_ALSA] alsa-lib: conf.c:4663:(snd_config_expand) Evaluate error: No such device [AO_ALSA] alsa-lib: pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM default [AO_ALSA] Playback open error: No such device Failed to initialize audio driver 'alsa' [AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le [AO SDL] using aalib audio driver. [AO SDL] Unable to open audio: No available audio device Failed to initialize audio driver 'sdl:aalib' Could not open/initialize audio device -> no sound. Audio: no sound and nothing changes in codec#0 file; with option -ao sdl, instead: [AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le ALSA lib confmisc.c:768:(parse_card) cannot find card '0' ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:4663:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM default AO: [sdl] 44100Hz 2ch s16le (2 bytes per sample) And sound works! Same changes as per (working) playback with kernel 33.2 (alsa 1.0.21) are noticeable in /proc/asound/card0/codec#0 in this case (that is, when getting sound with kernel 2.6.38.x); in addition, node 0x0c, now changes as well (again, '[Audio Output] wcaps 0x41d: Stereo Amp-Out.' and 'Converter: stream=5, channel=0') Moreover: with kernel 38.2, 38.3 alsamixer fails telling about problems finding device: cannot open mixer: No such device And amixer says: amixer: Mixer attach default error: No such device But there's no problem with /dev stuff (and same devices works pretty fine with kernel 2.6.33.2) - aplay -l doesn't work as well: aplay: device_list:240: no soundcards found... With kernel 2.6.33.2, aplay -l says: **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT2020 Digital [VT2020 Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 A 'third party' mixer embedded in my distribution aborts by hitting a failed assertion in function 'snd_hctl_load' (hcontrol.c) failing assertion being 'assert(hctl->count == 0)' - I've added a few simple lines above the assertion to print out some info on stderr, and got the following results: In kernel 2.6.33.2: hctl->count: 0 hctl->ctl: 134748144 (0x80817F0) hctl->ctl->name: hw:0 which seems reasonable - according to /proc/iomem address 0x80817F0 is in range 00100000-cf83ffff (system ram) and higher than kernel stuff subranges ( 01000000-012c3be7: Kernel code - 012c3be8-0141dadf : Kernel data - 01481000-014b1e03 : Kernel bss); In kernel 2.6.38.2 & .38.3: hctl->count: 1156 hctl->ctl: 5398 (0x1516) Segmentation fault suggesting memory corruption (that's always the same address) - according to /proc/iomem address 0x1516 is in first range (0x0000-0xffff) which is reserved. I thought (and hoped) that could have been somehow related to cached/noncached pages issue solved by commit c27b92295ab4c6b90b1cee94c4c9c1b4732e1c2e, but applying diff patch for kernel 2.6.38.3 didn't change anything. After that, I've read changelog for kermel 2.6.38.4 but such didn't seem to me to present related material, so I didn't updgrade. I hope this post can help solving this issue and improving alsa infrastracture (I cannot easily go further). Best regards.

2 19

[alsa-devel] Issues with HDSPM
by Andre Schramm 20 Jun '11

20 Jun '11

Hello, the HDSPM driver seems to be broken in the current ALSA release. We're using CentOS 5 with 2.6.33.9-rt31.64.el5rt and ALSA 1.0.24 compiled from source, the card is a PCIe Hammerfall DSP MADI (rev d2). As already mentioned in [1], the card operates in master mode, but the system clock mode can't be set to anything higher than 48kHz. # amixer cget numid=3 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=0 # amixer cset numid=2 5 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=5 # amixer cget numid=6 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=48000 When switching to a PCI version of the card, the kernel completely crashes on boot. And with [2], there is another issue which unfortunately hasn't been merged yet and still exists in the current release. Regards, Andre [1] http://sourceforge.net/mailarchive/message.php?msg_id=27520432 [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html

3 9

[alsa-devel] SPDIF In on ALC889 to speakers?
by Ady Deac 20 Jun '11

20 Jun '11

Hi guys, I have an GA-890FXA-UD5 mobo with an intel-hda sound card. I've purchased an additional SPDIF In adapter (bracket) to be able to play digital content on my 5.1 speakers. I have the speakers connected to the jacks (ie. not to the SPDIF Out). I've checked the Windows drivers, but these also only support bypassing SPDIF In to SPDIF Out. Is there any way to have this done by the card itself? Maybe some developers tweaks? Some chipset routing maybe? :D I know that the CM106 chipset needs some special initialization (eg: wr 00 20 05 30 01) to actually bypass the SPDIF In to the speakers, but it works! Anything that can be done in this case? Else, is there an easy way of bypassing the SPDIF Input to the speakers but also to allow the use of the card for playback? Thank you! -- Deac Mihai-Adrian W: www.mikesoftware.com P: +40-745-256.364, +40-723-85.44.44 F: +40-264-566.033 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40) # cat /proc/asound/card0/codec* Codec: Realtek ALC889 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0889 Subsystem Id: 0x1458a102 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=2 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=4 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=6 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x13 0x13] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 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 [0x570]: 32000 44100 48000 96000 192000 bits [0xe]: 16 20 24 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] [0x1f 0x1f] [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=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] 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: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] 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: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] 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: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19c50: [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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19c60: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x6, 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181345f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf 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: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, 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 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4005e601: [N/A] Line Out at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000020: IN Pin Default 0x01cb7170: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=28 Processing Coefficient: 0x00 Coefficient Index: 0x0f Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 12 0x18* 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 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] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x25 0x0b

1 0

[PATCH] ALSA: HDA: Remove quirk for an HP device
by David Henningsson 20 Jun '11

20 Jun '11

The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. BugLink: http://bugs.launchpad.net/bugs/761022 Cc: stable(a)kernel.org (v2.6.38+) Reported-by: Jo Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b97af9..c174865 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4884,7 +4884,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST), SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), -- 1.7.4.1 --------------080401020001090605070601--

1 0

[alsa-devel] [PATCH 3/3] ALSA: VIA HDA: Create a master amplifier control for VT1718S.
by lydiawang 20 Jun '11

20 Jun '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Create a master amplifier control for VT1718S. Create a master volume and mute control of playback for VT1718S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2011-06-20 13:49:00.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2011-06-20 14:01:15.000000000 +0800 @@ -4465,6 +4465,19 @@ if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT) { + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; /* Front */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(

1 0

[alsa-devel] [PATCH 2/3] ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S.
by lydiawang 20 Jun '11

20 Jun '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S. When switch HP independent mode, mute/unmute connctions of mixer which is connected to headphone for VT1718S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2011-06-20 13:41:04.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2011-06-20 13:48:07.000000000 +0800 @@ -745,12 +745,23 @@ struct via_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value; unsigned int pinsel = ucontrol->value.enumerated.item[0]; + unsigned int parm0, parm1; /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - if (spec->codec_type == VT1718S) + if (spec->codec_type == VT1718S) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + /* Set correct mute switch for MW3 */ + parm0 = spec->hp_independent_mode ? + AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0); + parm1 = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm0); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm1); + } else snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel);

1 0

[alsa-devel] [PATCH] ASoC: Fix DAPM sequence run for per-widget I/O methods
by Mark Brown 20 Jun '11

20 Jun '11

Previously we were using the DAPM context rather than a widget as the argument for update_bits() so we didn't need to care that our list walk of widgets left us one beyond the end of the list. Now we're using them for the register update we need to make sure we're pointing at an actual widget not the list_head. Fix originally suggested by Liam on IM. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 605c225..ceb2ba4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -918,6 +918,12 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, } if (reg >= 0) { + /* Any widget will do, they should all be updating the + * same register. + */ + w = list_first_entry(pending, struct snd_soc_dapm_widget, + power_list); + pop_dbg(dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); -- 1.7.5.4

2 1