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[alsa-devel] [Fwd: Re: [regression] 2.6.25-rc4 snd-es18xx broken on Alpha]
by Tyson Whitehead 02 Apr '08

02 Apr '08

Sorry. Bob was just saying this could be an important observation, but it didn't get to everyone as the CC list got shortened. Cheers! -Tyson

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Re: [alsa-devel] [regression] 2.6.25-rc4 snd-es18xx broken on Alpha
by rct＠frus.com 02 Apr '08

02 Apr '08

rct wrote: > Rene Herman wrote: > > Okay, and applying the attached just makes your sound completely dead in the > > water? > > (patch to remove es1888_init from a Miata build omitted) > > I'll try a build with the old OSS "sb" driver, and if that works ok, we > may be able to do away with es1888_init() on the Miata. Tyson -- I > think you have a Miata if I'm remembering correctly: can you confirm > these observations? Quick followup: OSS "sb" driver works fine without es1888_init(). -- ------------------------------------------------------------------------ Bob Tracy | "I was a beta tester for dirt. They never did rct(a)frus.com | get all the bugs out." - Steve McGrew on /. ------------------------------------------------------------------------

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[alsa-devel] How to reduce delay between playback and capture for echo canceller
by Cliff Cai 01 Apr '08

01 Apr '08

Hi all, I'm playing a linphone based VOIP demo board these days,the board can connect different soundcard daughter board as playback and capture devices through I2S or AC97.Then I experience a bad echo on my PC(using liphone and headset) when talk with the board (with speaker and MIC on board).So I enable the echo canceller (based on speex) for it. One of my sound card (connected by I2S) works pretty well,but the other one (connected by AC97) doesn't work. Through the diagnostic tool provided by speex I find that the ac97 one has a big delay between played signal(reference signal) and captured signal.Just the delay make echo canceller don't work as expectation.So I think the driver for the AC97 soundcard need some improvement to reduce the delay. Has anyone a usefull direction for me? Thanks in advance,Best Regards Cliff Cai _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

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[alsa-devel] [Fwd: Re: Davinci ASoC support (patch)]
by lokowich 01 Apr '08

01 Apr '08

Forwarded from davinci-linux-open-source list: I've built this ASoC patch into my DaVinci git kernel (thanks for the SENSORS tip), and can use aplay and speaker-test, but no sound. Figured it may be a mixer/volume control issue, but the alsamixer is unable to open for a failed ioctl on USBDEVFS_IOCTL. Not sure why there is a USB dependency. Noted that there is no /sys/class/sound/mixer in my sysfs. Any ideas? Mark Bernard Blackham wrote: > Behme wrote: > >> While searching something different, I just found >> >> http://www.spinics.net/lists/alsa-devel/msg12535.html >> >> As I'm not subscribed to alsa-devel, I don't have the original post. >> Maybe any of the authors can post it do this list as well? >> >> The patch mainly applies to our git. I fixed two small rejects in >> Kconfig and Makefile as the patch is for newer kernel than our git. >> Untested patch against our git in attachment. Anybody likes to test? >> > > Well, I tried compiling it and it failed to build. It appears that it's > built on a set of patches for Davinci posted to LAK, including this > completely revamped EDMA support[1]: > > http://www.spinics.net/lists/arm-kernel/msg47450.html > > So I had to change a few definitions, import a few functions from the new > DMA support and managed to get it compiling and loading, but not yet working > (tried various sample rates and formats): > > root@davinci-dvevm:/tst$ aplay test.wav > Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable > aplay: set_params:878: Broken configuration for this PCM: no configurations > available > > root@davinci-dvevm:/tst$ arecord -r 48000 -f S16_LE > Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono > ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable > arecord: set_params:878: Broken configuration for this PCM: no > configurations available > > > And this is the extent of my knowledge of ALSA - I can continue digging, but > maybe somebody else will beat me to it =) It's looking promising though. > > Attached is the patch I have so far that applies against current git. This > is not for going upstream, but just for testing. Notes: > - Apply patch > - Enable SND_SOC, SND_DAVINCI_SOC, SND_DAVINCI_SOC_I2S, SND_DAVINCI_SOC_EVM > and SND_SOC_TLV320AIC3X. > - Make sure CONFIG_I2C_DAVINCI is enabled, but disable the > CONFIG_SENSORS_TLV320* options. > - Build. > - Load the snd-soc-evm module. > > Bernard. > > [1] These patches appear to have gone absolutely nowhere beyond that list. I > can't find them in any tree. But they are cleaner and more conforming to the > Linux Kernel style than what's currently in git. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Davinci-linux-open-source mailing list > Davinci-linux-open-source(a)linux.davincidsp.com > http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source >

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[alsa-devel] Keep getting underruns
by Barclay, Bruce 31 Mar '08

31 Mar '08

Not sure what causes this in the Alsa framework. I only get one at the beginning of the song but not sure how and why I get it. Does it have to do with the return of my pointer callback or because I call snd_period_elapsed in my interrupt routine too soon. Any ideas or docs to read? Bruce

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[alsa-devel] M-Audio Revolution 7.1 sound card causing crash on Alpha
by Michael Cree 31 Mar '08

31 Mar '08

I recently installed a M-Audio Revolution 7.1 sound card into DEC/Compaq Alpha XP1000, and it is causing crashes and complete lock ups of the computer when I attempt to play sound through the sound card. Since this is my main computer I shifted the card to a DEC Alpha PWS500au and it causes the same lock ups there, and in one case a kernel oops made it into the system log (listed below). The crashes occur when any application plays sound through the card. The first instance of running an application (such as aplay or speaker-test) after bootup succeeds; it's the second time one runs such an application that the system crashes. I am using the ice1724 alsa driver. The sound card works fine - I've verified it on a MS Windows machine. It also worked fine when I installed it in a x86_64 machine running Debian testing. This seems to be a bug possibly specific to the Alpha architecture. Any ideas as to the problem? Any suggestions as to effective debugging? I turned on the debug options for alsa in the kernel config and re-compiled, but couldn't find any alsa debugging information in any of the system logs. Basic system details follow: DEC Alpha PWS500au with a 600MHz Alpha EV56 cpu. Debian GNU/Linux - up to date testing (Lenny) distribution. 2.6.24.3 verbatim kernel self-compiled and alsa as provided in that kernel. The kernel oops from one crash is as follows: Mar 4 22:04:04 aleph kernel: Kernel bug at mm/mmap.c:2054 Mar 4 22:04:04 aleph kernel: speaker-test(1885): Kernel Bug 1 Mar 4 22:04:04 aleph kernel: pc = [<fffffc000036bec4>] ra = [<fffffc000036be88>] ps = 0000 Not tainted Mar 4 22:04:04 aleph kernel: pc is at exit_mmap+0x134/0x150 Mar 4 22:04:04 aleph kernel: ra is at exit_mmap+0xf8/0x150 Mar 4 22:04:04 aleph kernel: v0 = 0000000000000000 t0 = 0000000000000003 t1 = 0000000000000040 Mar 4 22:04:04 aleph kernel: t2 = 000000000000003f t3 = fffffc002300c5f8 t4 = 0000000000000001 Mar 4 22:04:04 aleph kernel: t5 = fffffc00001da000 t6 = 0000000000000000 t7 = fffffc00212dc000 Mar 4 22:04:04 aleph kernel: a0 = 0000000000000000 a1 = fffffc002300c400 a2 = 0000000000000000 Mar 4 22:04:04 aleph kernel: a3 = 0000000000000000 a4 = 0000000000000000 a5 = 0000000000000000 Mar 4 22:04:04 aleph kernel: t8 = 0000000000000000 t9 = 0000000d1bc1e51d t10= 1a00000000000000 Mar 4 22:04:04 aleph kernel: t11= 0000000000002000 pv = fffffc000037b3d0 at = 0000000000000003 Mar 4 22:04:04 aleph kernel: gp = fffffc00006fb518 sp = fffffc00212dfbe8 Mar 4 22:04:04 aleph kernel: Trace: Mar 4 22:04:04 aleph kernel: [<fffffc000032516c>] mmput+0x5c/0x100 Mar 4 22:04:04 aleph kernel: [<fffffc0000329820>] exit_mm+0xc0/0x180 Mar 4 22:04:04 aleph kernel: [<fffffc000032a95c>] do_exit+0x16c/0x950 Mar 4 22:04:04 aleph kernel: [<fffffc000032b184>] do_group_exit+0x44/0xc0 Mar 4 22:04:04 aleph kernel: [<fffffc0000335a9c>] get_signal_to_deliver+0x2fc/0x450 Mar 4 22:04:04 aleph kernel: [<fffffc00003167f4>] do_notify_resume+0xb4/0x570 Mar 4 22:04:04 aleph kernel: [<fffffc00003110cc>] work_pending+0x5c/0x70 Mar 4 22:04:04 aleph kernel: [<fffffc0000333f60>] __sigqueue_alloc+0x40/0xc0 Mar 4 22:04:04 aleph kernel: [<fffffc000038f7a0>] do_ioctl+0x30/0x90 Mar 4 22:04:04 aleph kernel: [<fffffc0000334954>] specific_send_sig_info+0xd4/0x110 Mar 4 22:04:04 aleph kernel: [<fffffc0000334a9c>] force_sig_info+0x8c/0xe0 Mar 4 22:04:04 aleph kernel: [<fffffc0000310bdc>] entMM+0x9c/0xc0 Mar 4 22:04:04 aleph kernel: Mar 4 22:04:04 aleph kernel: Code: a77df1b0 6b5b497f 27ba0039 23bdf664 c3ffffec 00000081 <00000806> 005efcfe Mar 4 22:04:04 aleph kernel: Fixing recursive fault but reboot is needed! Modules installed: ipv6 evdev snd_ice1724 snd_ice17xx_ak4xxx snd_ac97_codec ac97_bus snd_ak4114 snd_pt2258 snd_i2c snd_ak4xxx_adda snd_mpu401_uart snd_pcm_oss snd_pcm snd_page_alloc snd_mixer_oss snd_seq_dummy snd_seq_oss snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device snd soundcore pcspkr serio_raw tulip Sound devices: **** List of PLAYBACK Hardware Devices **** card 0: Revolution71 [M Audio Revolution-7.1], device 0: ICE1724 [ICE1724] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Revolution71 [M Audio Revolution-7.1], device 1: IEC1724 IEC958 [IEC1724 IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Revolution71 [M Audio Revolution-7.1], device 2: ICE1724 Surrounds [ICE1724 Surround PCM] Subdevices: 3/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 cat /proc/asound/cards: 0 [Revolution71 ]: ICE1724 - M Audio Revolution-7.1 M Audio Revolution-7.1 at 0x8480, irq 40 PCI devices: 00:03.0 Ethernet controller: Digital Equipment Corporation DECchip 21142/43 (rev 30) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 255 (5000ns min, 10000ns max), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 24 Region 0: I/O ports at 9000 [size=128] Region 1: Memory at 09259000 (32-bit, non-prefetchable) [size=128] Expansion ROM at 09200000 [disabled] [size=256K] Kernel driver in use: tulip Kernel modules: tulip 00:07.0 ISA bridge: Contaq Microsystems 82c693 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 00:07.1 IDE interface: Contaq Microsystems 82c693 (prog-if 80 [Master]) Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 0 Region 0: I/O ports at 01f0 [size=8] Region 1: I/O ports at 03f4 [size=1] Region 2: [virtual] Memory at 00000170 (32-bit, non-prefetchable) [disabled] [size=8] Region 3: [virtual] Memory at 00000370 (type 3, non-prefetchable) [disabled] [size=1] Region 4: I/O ports at 9080 [size=16] Kernel driver in use: Cypress_IDE 00:07.2 IDE interface: Contaq Microsystems 82c693 (prog-if 00 []) Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 0 Region 0: I/O ports at 01f0 [size=8] Region 1: I/O ports at 03f4 [size=1] Region 2: [virtual] Memory at 00000170 (32-bit, non-prefetchable) [disabled] [size=8] Region 3: [virtual] Memory at 00000370 (type 3, non-prefetchable) [disabled] [size=1] Region 4: Memory at 09240000 (32-bit, non-prefetchable) [disabled] [size=64K] 00:07.3 USB Controller: Contaq Microsystems 82c693 (prog-if 10 [OHCI]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 248 Interrupt: pin A routed to IRQ 10 Region 0: Memory at 09258000 (32-bit, non-prefetchable) [size=4K] Kernel driver in use: ohci_hcd 00:0c.0 VGA compatible controller: Digital Equipment Corporation PBXGB [TGA2] (rev 22) (prog-if 00 [VGA controller]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR+ INTx- Latency: 255 (2000ns min, 16000ns max) Interrupt: pin A routed to IRQ 32 Region 0: Memory at 10000000 (32-bit, prefetchable) [size=128M] Expansion ROM at 09250000 [disabled] [size=32K] 00:14.0 PCI bridge: Digital Equipment Corporation DECchip 21152 (rev 02) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 255, Cache Line Size: 64 bytes Bus: primary=00, secondary=01, subordinate=01, sec-latency=255 I/O behind bridge: 00008000-00008fff Memory behind bridge: 09000000-090fffff Prefetchable memory behind bridge: 0000000009100000-00000000091fffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- 01:04.0 SCSI storage controller: QLogic Corp. ISP1020 Fast-wide SCSI (rev 05) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 248, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 27 Region 0: I/O ports at 8000 [size=256] Region 1: Memory at 09000000 (32-bit, non-prefetchable) [size=4K] Expansion ROM at 09100000 [disabled] [size=64K] Kernel driver in use: qla1280 01:09.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: VIA Technologies Inc. M-Audio Revolution 7.1 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 248 Interrupt: pin A routed to IRQ 40 Region 0: I/O ports at 8480 [size=32] Region 1: I/O ports at 8400 [size=128] Capabilities: [80] Power Management version 1 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: ICE1724 Kernel modules: snd-ice1724 Cheerz Michael.

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[alsa-devel] [Fwd: Re: Davinci ASoC support (patch)]
by lokowich 31 Mar '08

31 Mar '08

Vova Barinov wrote: > Mark Brown wrote: > >> On Fri, Mar 28, 2008 at 09:27:04AM -0500, lokowich wrote: >> >> >> >>> Forwarded from davinci-linux-open-source list: >>> >>> >> CCing in Vladimir Barinov who contributed the DaVinci support. >> >> >> >>> I've built this ASoC patch into my DaVinci git kernel (thanks for the >>> SENSORS tip), and can use aplay and speaker-test, but no sound. Figured >>> it may be a mixer/volume control issue, but the alsamixer is unable to >>> open for a failed ioctl on USBDEVFS_IOCTL. Not sure why there is a USB >>> dependency. Noted that there is no /sys/class/sound/mixer in my sysfs. >>> Any ideas? >>> >>> > How do you figure out that you can play sound? > Also what platform do you use - is it Davinci-EVM? what line output do > you use to hear the sound? > Please, post your dmesg log and cat /proc/interrupts too. > Excerpt from dmesg: Advanced Linux Sound Architecture Driver Version 1.0.16. ASoC version 0.13.2 AIC3X Audio Codec 0.1 asoc: aic3x <-> davinci-i2s mapping ok ALSA device list: #0: DaVinci EVM (aic3x) /proc/interrupts: CPU0 6: 0 AINTC dm644xresizer 8: 23608 AINTC davincifb 13: 6995 AINTC EMAC 22: 107 AINTC ide0 32: 39436 AINTC clockevent 33: 3 AINTC free-run counter 39: 217 AINTC i2c_davinci 40: 204 AINTC serial Err: 0 This is the DM6446 EVM. Note, my kernel has been patched with latest ALSA, to no avail. I hear the DAC pop during initialization on both the Line Out and Headphone jacks. >> It might be helpful to post dmesg output showing the sound support >> being loaded >> >> >>> Bernard Blackham wrote: >>> >>> >>>> Behme wrote: >>>> >>>> >>>> >>>>> While searching something different, I just found >>>>> >>>>> http://www.spinics.net/lists/alsa-devel/msg12535.html >>>>> >>>>> As I'm not subscribed to alsa-devel, I don't have the original post. >>>>> Maybe any of the authors can post it do this list as well? >>>>> >>>>> The patch mainly applies to our git. I fixed two small rejects in >>>>> Kconfig and Makefile as the patch is for newer kernel than our git. >>>>> Untested patch against our git in attachment. Anybody likes to test? >>>>> >>>>> >>>>> >>>> Well, I tried compiling it and it failed to build. It appears that it's >>>> built on a set of patches for Davinci posted to LAK, including this >>>> completely revamped EDMA support[1]: >>>> >>>> http://www.spinics.net/lists/arm-kernel/msg47450.html >>>> >>>> > Yep, it's needed. > >>>> So I had to change a few definitions, import a few functions from the new >>>> DMA support and managed to get it compiling and loading, but not yet working >>>> (tried various sample rates and formats): >>>> >>>> root@davinci-dvevm:/tst$ aplay test.wav >>>> Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo >>>> ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable >>>> aplay: set_params:878: Broken configuration for this PCM: no configurations >>>> available >>>> >>>> root@davinci-dvevm:/tst$ arecord -r 48000 -f S16_LE >>>> Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono >>>> ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable >>>> arecord: set_params:878: Broken configuration for this PCM: no >>>> configurations available >>>> >>>> >>>> And this is the extent of my knowledge of ALSA - I can continue digging, but >>>> maybe somebody else will beat me to it =) It's looking promising though. >>>> >>>> Attached is the patch I have so far that applies against current git. This >>>> is not for going upstream, but just for testing. Notes: >>>> - Apply patch >>>> - Enable SND_SOC, SND_DAVINCI_SOC, SND_DAVINCI_SOC_I2S, SND_DAVINCI_SOC_EVM >>>> and SND_SOC_TLV320AIC3X. >>>> - Make sure CONFIG_I2C_DAVINCI is enabled, but disable the >>>> CONFIG_SENSORS_TLV320* options. >>>> - Build. >>>> - Load the snd-soc-evm module. >>>> >>>> Bernard. >>>> >>>> [1] These patches appear to have gone absolutely nowhere beyond that list. I >>>> can't find them in any tree. But they are cleaner and more conforming to the >>>> Linux Kernel style than what's currently in git. >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Davinci-linux-open-source mailing list >>>> Davinci-linux-open-source(a)linux.davincidsp.com >>>> http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source >>>> >>>> >>>> >>> _______________________________________________ >>> Alsa-devel mailing list >>> Alsa-devel(a)alsa-project.org >>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>> >>> >>> >> >> > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

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Re: [alsa-devel] Any wigets for ADC to DAC ?
by Mark Brown 31 Mar '08

31 Mar '08

On Mon, Mar 31, 2008 at 12:32:21PM +0200, Alexandre BOUIN wrote: > Will be there a support for a such feature in ASoC v2 ? Given that more and more codecs support this sort of thing it's likely that support for digital paths will get implemented at some point. No specific ETA for it, though. Note that if there is no specific reason to use a digital bypass path (such as filtering that is only supported on the digital path or a lack of a suitable analogue path) then it is normally more power efficient to use an analogue bypass path than a digital one.

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Re: [alsa-devel] Any wigets for ADC to DAC ?
by Mark Brown 31 Mar '08

31 Mar '08

On Fri, Mar 28, 2008 at 04:39:48PM +0100, Alexandre BOUIN wrote: > In intern, we're still using WM8900 codec. We'd like to know in ASoC layer, > there is a widget to declare a possible audio path between ADC and DAC. If there were a direct connection the appropriate thing would just be to declare it (in that particular chip there is actually a digitial mixer in there). Using such a path is not particularly supported by ASoC v1 since it treats ADCs and DACs as sinks and sources only.

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[alsa-devel] [PATCH]Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880
by Jiang zhe 31 Mar '08

31 Mar '08

Hi, To mute the output of Pin widget 15 in ALC880, we should use the HDA_OUTPUT. However, current code looks like : snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); It may be a misspelling. Best regards! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

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