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[alsa-devel] Merging hg sources into kernel tree
by Andrew Boettcher 20 Feb '08

20 Feb '08

Is there a script to merge the hg sources into a kernel tree? If not do I need to worry about any changes other than the include/ directory? I have another module (cx88-alsa) that is in v4l but needs the newer alsa tree or I get "cx88_alsa: disagrees about version of symbol xxx", so I have to merge the sources for now. Andrew Boettcher

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[alsa-devel] Moving X Window windows causes audio stuttering
by Timur Tabi 20 Feb '08

20 Feb '08

On our system, using mplayer to play a movie that has 44.1KHz audio (hardware only supports 48KHz) can cause audio stuttering if we're using OSS emulation, but not if we're using native ALSA. Is this a known problem? Does OSS emulation hog the CPU? -- Timur Tabi Linux kernel developer at Freescale

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Re: [alsa-devel] [PATCH] Driver for Emagic Audiowerk 2
by Takashi Iwai 20 Feb '08

20 Feb '08

At Tue, 19 Feb 2008 00:38:54 +0100, Cédric Brégardis wrote: > > Signed-off-by: Cedric Bregardis <cedric.bregardis(a)free.fr> > > Signed-off-by: Jean-Christian Hassler <jhassler(a)free.fr> Could you avoid HTML mail? I cannot get the patch properly. If embedding text is difficult, rather use an attachment. thanks, Takashi

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[alsa-devel] Ends up with write error
by Ning Zhang 20 Feb '08

20 Feb '08

Hi all, I am writing a ALSA driver for my ARM 926 platform. The ARM communicates with the DSP by accessing a 128-byte shared memory. I don't use IRQ mechanism.(In fact, I am not sure the platform has this mechanism) Here is some code slices: static struct snd_pcm_hardware snd_arch_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000), .rate_min = 8000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 128, .period_bytes_min = 2, .period_bytes_max = 128, .periods_min = 1, .periods_max = 64, }; static snd_pcm_uframes_t snd_arch_pcm_pointer(struct snd_pcm_substream *substream) { … /* * get the current hardware pointer */ current_ptr = arch_fw_read_dsp_reg(chip, AFE_RWADDR) & AFE_RD_ADDR; return bytes_to_frames(runtime, current_ptr); } static int snd_arch_pcm_playback_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count) { …… count = frames_to_bytes(runtime, count); pos = frames_to_bytes(runtime, pos); chip->bufpos = pos; …. spin_lock(chip->reg_lock); if ( count > 0 && count <= MAX_BUF_SIZE ) { if ( copy_from_user(chip->bufptr, src, count) ) { snd_printk(KERN_ERR "can not copy user data!\n"); return -EFAULT; } if ( arch_fill_rx_buf(chip, (signed short*)(chip->bufptr), count) < 0 ) { snd_printk(KERN_ERR "fill rx buffer failed!\n"); return -1; } …………… } spin_unlock(chip->reg_lock); return 0; } The problem is my driver can make sounds, but ends up a write error. After much debugging, I think there might be some problem in the interworking between middle-level driver and the device driver. I found the runtime->control->avail_min=32, and the runtime->buffer_size and runtime->period_size are both 32. So at first middle-level driver call snd_pcm_lib_write1 transfers 32 frames(128 bytes) to the device driver's copy callback function and trigger function. Then when middle-level driver wants to transfer another 32 frames, it calls the pointer callback function and gets the position, because at this time the whole bytes has not been processed completely, so the pointer function's return value must less than 32, and the avail must less than runtime->control_->avail_min, and goes to this flow and ends up a write error. _______________________________________________________________________________ if (((avail < runtime->control->avail_min && size > avail) || (size >= runtime->xfer_align && avail < runtime->xfer_align))) { ……………... long tout; init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); while (1) { if (signal_pending(current)) { state = SIGNALED; break; } set_current_state(TASK_INTERRUPTIBLE); snd_pcm_stream_unlock_irq(substream); tout = schedule_timeout(10 * HZ); snd_pcm_stream_lock_irq(substream); if (tout == 0) { if (runtime->status->state != SNDRV_PCM_STATE_PREPARED && runtime->status->state != SNDRV_PCM_STATE_PAUSED) { state = runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED; break; } } switch (runtime->status->state) { ………….. : state = ERROR; goto _end_loop; default: break; } avail = snd_pcm_playback_avail(runtime); if (avail >= runtime->control->avail_min) { state = READY; break; } } _end_loop: remove_wait_queue(&runtime->sleep, &wait); switch (state) { ……. case EXPIRED: snd_printd("playback write error (DMA or IRQ trouble?)\n"); err = -EIO; goto _end_unlock; ……… } } This code is a part of snd_pcm_lib_write1() in sound/core/pcm_lib.c I think that I misunderstood the mechanism of middle-level driver. Could you please correct me? What should the pointer callback function return? What's it for? How comes the runtime->control->avail_min? Because I don't use the IRQ, so should I use the snd_pcm_period_elapsed function? And where? ALSA Version is 1.0.14rc1 and I use alsa-lib-1.0.15. Can someone comment on this and give me some hints to solve this problem? Thanks in advance. BRs, David

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[alsa-devel] ALSA MIDI interface - plugin
by Pharaoh . 20 Feb '08

20 Feb '08

Hi I am writing an user space ALSA i/o type PCM plugin for my board. I chose this rather unconventional approach for supporting ALSA apps since the the the driver which communicates with DSP using shared memory is in userspace and for some constraints the driver communication driver will always be in userspace. My audio plugin uses calls of the coomunication driver and talks to DSP. I already have a working plugin in place. I use it like following, #aplay -D my_plugin test.wav and it works. Now, I have to support rawmidi interface also. Is it possible to write a userspace plugin for making MIDI work? Where can I get sample code etc? -pharaoh.

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Re: [alsa-devel] Problem with setting period time and buffer time.
by Clemens Ladisch 20 Feb '08

20 Feb '08

Ignacy Kasperowicz wrote: > But what about buffer_time, why can't I set this time ? The allowed range of values is hardware dependent, like the period time. Use snd_pcm_hw_params_set_buffer_time_near(). > Anyway is there a method to play my wave for a specified short time? Just write the data to the device. What is the problem? Regards, Clemens

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[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
by Daniele (Mastro) 19 Feb '08

19 Feb '08

hi all i'm new to this mailing list i've bought a laptop and i would like to help you supporting my sound card what works: 2 speaker, low volume (it should be a 5+1 audio card) what doesn't -) low volume -) inoperating headphone jack -) not working mic -) no subwoofer -) may be some other stuff not tested ( SPDIF? hardware midi?) with speaker-test -t wav -c 6 only front left/right channel works my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux alsa 1.0.15 drivers (compiled with debian-way: module-assistant a-i alsa-source) i've already tried all the option i could think about auto (default) toshiba 3stacks 3stacks-6ch 3stacks-6ch-dig neither one has better/worst behavior I've checked the module has toke those option by cat /sys/module/snd_hda_intel/parameters/model anyway I've read the ALSA-Configuration.txt shipped with kernel and for Codec ALC268 it only has 4 options ALC268 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops auto auto-config reading BIOS (default) I've googled a bit and i found out that the Toshiba A205 has completely different problem.. the sound doesn't work with it.. reading the changelog from 1.0.14 to 1.0.15 (alsa) it say: Added the proper model=toshiba for Toshiba A305 with ALC268 codec. so i think my audio card is an unknown model the laptop is brand new and i only found this model in Italy anyway i think that the same audio card is shipped with the Toshiba Satellite x205-S7483 witch is an american laptop some info about the audio card: lspci -n 00:1b.0 0403: 8086:284b (rev 03) lspci -vvv 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f6600000 (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] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [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] Connection: 2 0x02 0x1d Node 0x10 [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] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: 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 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f 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 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13 many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device i've already filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659 please ask me if you need any more info or want me to do some test thanks you all for your work ps: i've subscribed to vger mailing list, what's the right mailing list for alsa-devel? -- Daniele Segato

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[alsa-devel] Maya44 revised patch
by Rainer Zimmermann 19 Feb '08

19 Feb '08

This is the revised version of the ESI Maya44 patch. changes are mainly coding style, some cleanups and using a patch file for ice1724.c . The added code now (mostly) passes checkpatch.pl, except for some "#if 0"'s which I'd leave in for the development phase. Also, I changed the rate setting logic, which (for now) allows all rates up to 192kHz even for capturing. As capturing actually only supports 96 kHz, while capturing & playback rates are linked, this is not really ok. I'd appreciate suggestions about how this could be handled properly, while still supporting playback up to 192 kHz... Again, feedback & testing is appreciated. For more information, see doc/README.maya44 . -Rainer -- Lightshed IT Services Löfflerstr. 27, 22765 Hamburg, Germany * mail(a)lightshed.de * www.lightshed.de

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[alsa-devel] snd-hda-intel AD1981 - Sound volume low level
by Robert Gonciarz 19 Feb '08

19 Feb '08

Hi, I just want to ask you if you have heard about the problem with low sound volume level even though you set the alsamixer on the (almost) maximum volume level. The sound card: snd-hda-intel / AD1981 / ICH7 ) It's a big problem for me especially when I watch a movie with ac3 (DVD quality) decoded sound. The volume level is extrimly low in such situation (I mean the level you hear the sound). I have to connect my loudspeakers to a ampilifier and set it in very high level. I don't have such problem when I work with Windows XP. They are loud enough. That's alsa-info.sh result: http://pastebin.ca/908831 It is said that there are some preferences in the totem package (hidden in alsamixer) that can fix it and make alsa to remember higher sound level. I've noticed that problem with all kernels I tested (I've checked about 15-20 kernels from that range 2.6.13.x - 2.6.24.x) (I usually use vanillia kernel (from www.kernel.org) /with build alsa/). But I've tested the newest alsa-drivers: 1.0.16 with 2.6.24.2 kernel also and the result was the same. Could you chack that problem, please? Robert

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Re: [alsa-devel] Problem with setting period time and buffer time.
by Ignacy Kasperowicz 19 Feb '08

19 Feb '08

Ok, so I misunderstood the idea of a period variable, many thanks for explanation Clemens. I've just read that buffer_time also isn't what I think it was ;) It's a size of the ring buffer of sound card and it's length depends on hardware, am I right? So the best way is to depend on default values of hardware? Anyway is there a method to play my wave for a specified short time? As I can see using timer interface is the only solution but still can't google anything useful :( Small example how to apply timers will be appreciated (I saw example timer.c from alsa-lib, but I'd prefer to read some explanation how the code works). Thanks from advance for any replays. -- ################################################ gry świetlicowe drogą do wyższej fazy przemian. gg:681548 jid: fik(a)jabbim.pl usecide.net ################################################

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