[Alsa-devel] ALSA for OS X ???
Hej, in order to convert some sound applikations to OS X I need ALSA for it. Is there anyone who ever tried running it this way or are there any ideas ? -- A l e x --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] PCM API enhancement - more bitmasks to support more formats
At Thu, 27 Jun 2002 00:08:34 +0200, Frank van de Pol wrote: Hi Jaroslav, you might be interested in next site: http://www.tsp.ece.mcgill.ca/Docs/AudioFormats/WAVE/WAVE.html wow, it's a good collection. thanks for this info! ciao, Takashi --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] 1869 trouble with snd-es18xx
Hi, I'm still trying to get ALSA running on my es1869 chip. The chip gets detected properly: ALSA ../alsa-kernel/isa/es18xx.c:1544: [0x220] ESS1869 chip found Also alsamixer works properly. But when I try to play a soundfile with alsaplayer I just get whitish noise. I get a lot of messages like this from dmesg: ALSA ../alsa-kernel/core/pcm_native.c:1093: playback drain error (DMA or IRQ trouble?) I tried other IRQ's and DMA's, and then the chip doesn't get recognised, or I don't get noise when playing something. Also, the IRQ and DMA settings (IRQ 5, DMA1 1, DMA2 5) are the same for the OSS sb driver, which is working properly. So... What can I do? I took a look at es18xx.c but my knowledge about device driver programming is very limited. And I'm longing to try ALSA, and JACK, and some other stuff... Is there a way to debug device drivers, maybe with gdb? Isn't there any option to use a 'virtual' ALSA device? Like snd-pcm-oss, but then the other way around. snd-oss-pcm module? Then you could use ALSA with a working OSS driver, and no working ALSA driver... greetz, Kasper --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] rc2 stereo problem with via8233
Hi, At Wed, 26 Jun 2002 16:44:58 -0400, John covici wrote: Here is what I get from the lspci -- I don't think that the via8233 is multi-channel at all and I didn't see any controls to utilize such a thing in my asound.state file. in fact via8233 has capability to output up to 6 channels (and spdif), although most of motherboards have only two output channels. the difference is only the implementation of the driver, so no change will appear on /etc/asound.state. the channels are configured dynamically according to the requested channels of pcm. 00:11.5 Multimedia audio controller: VIA Technologies, Inc. AC97 Audio Controller (rev 30) I can send you the verbose version if that would help. yes, could you get more detailed info with -xvv and -nvv options? and, please recheck under which condition this happens, whether with the stereo samples or with the mono samples, or whatever. i can revert the codes only for certain chip models. since via8233 has _only_ multi-channel playback mode, it must use this. but other chip models have another mode, which was used by the old driver. this can be enabled conditionally by checking the chip revision number. Takashi --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] changed snd_pcm_close() behaviour
On Wed, 26 Jun 2002, Abramo Bagnara wrote: Jaroslav Kysela wrote: Hi all, after discussion with Takashi and others on this list, the behaviour of snd_pcm_close() function has been changed in this way: The stream is stoped and all samples (if any) are droped. It means, that if the user space code wants to flush samples to output, the snd_pcm_drain() function must be called before the snd_pcm_close() call. I hope that this change doesn't cause any problems in any application. Anyway, please, report all problems. Please don't forget to fix alsa-oss. Done. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project http://www.alsa-project.org SuSE Linuxhttp://www.suse.com --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] rc2 stereo problem with via8233
OK, here is -xvv for that controller. 00:11.5 Multimedia audio controller: VIA Technologies, Inc. AC97 Audio Controller (rev 30) Subsystem: VIA Technologies, Inc.: Unknown device 4511 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin C routed to IRQ 11 Region 0: I/O ports at ec00 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 00: 06 11 59 30 01 00 10 02 30 00 01 04 00 00 00 00 10: 01 ec 00 00 00 00 00 00 00 00 00 00 00 00 00 00 20: 00 00 00 00 00 00 00 00 00 00 00 00 06 11 11 45 30: 00 00 00 00 c0 00 00 00 00 00 00 00 0b 03 00 00 and here is the -nvv 00:11.5 Class 0401: 1106:3059 (rev 30) Subsystem: 1106:4511 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin C routed to IRQ 11 Region 0: I/O ports at ec00 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- When I was using it, a stereo sample did play on both channels, from the cd, from pcm, they were mono -- I didn't test with stereo for the pcm. Is this sound card a multi channel and if so, how do I use more than one channel? on Thursday 06/27/2002 Takashi Iwai([EMAIL PROTECTED]) wrote Hi, At Wed, 26 Jun 2002 16:44:58 -0400, John covici wrote: Here is what I get from the lspci -- I don't think that the via8233 is multi-channel at all and I didn't see any controls to utilize such a thing in my asound.state file. in fact via8233 has capability to output up to 6 channels (and spdif), although most of motherboards have only two output channels. the difference is only the implementation of the driver, so no change will appear on /etc/asound.state. the channels are configured dynamically according to the requested channels of pcm. 00:11.5 Multimedia audio controller: VIA Technologies, Inc. AC97 Audio Controller (rev 30) I can send you the verbose version if that would help. yes, could you get more detailed info with -xvv and -nvv options? and, please recheck under which condition this happens, whether with the stereo samples or with the mono samples, or whatever. i can revert the codes only for certain chip models. since via8233 has _only_ multi-channel playback mode, it must use this. but other chip models have another mode, which was used by the old driver. this can be enabled conditionally by checking the chip revision number. Takashi -- John Covici [EMAIL PROTECTED] --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] 1869 trouble with snd-es18xx
Paul Davis wrote: Isn't there any option to use a 'virtual' ALSA device? Like snd-pcm-oss, but then the other way around. snd-oss-pcm module? Then you could use ALSA with a working OSS driver, and no working ALSA driver... ALSA has a significantly different internal architecture than OSS. It is not possible to use an OSS driver to support the ALSA midlevel code. What kind of midlevel code? I meant an ALSA driver, that you can assign an OSS device. The midlevel ALSA stuff can be handled by the ALSA driver, that just sends the audio data to the OSS device. greetz, Kasper --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Midi reverb on Emu10k1
On Thu, 27 Jun 2002, Rui Sousa wrote: On Wed, 26 Jun 2002, Jaroslav Kysela wrote: On Wed, 26 Jun 2002, Takashi Iwai wrote: Hi, At Mon, 24 Jun 2002 20:38:00 +0200, Nicola Orru' wrote: qq Hello, people out there! I'm Yet-Another-Programmer-Keyboard player. I would like to work on MIDI chorus/reverb implementations on the emu10k1 driver in a way I can experience as much fun as original Creative driver provide on Windogs. Where should I start from? How much work is needed to complete the sub-project? Where can I find documentation about that? this is one of long standing todo's. the implementation needs to steps. 1. write a dsp loader program to communicate with the alsa-drivers via emu10k1's hwdep ioctls. there is already an assembler, which was ported from oss driver's program. but it's never tested, i think. Well, the assembler probably needs more modifications, because my idea of DSP code management and linking is different from the OSS guys. 2. adds a code to assign the dsp codes to certain midi controls in emu10k1 driver. once the first one is ready, then it wouldn't take long time to do the 2nd step. i have already some idea about the latter. Currently, I am trying to learn more about DSP architectures and digital filter programming on several online books, but, as I have little patience, I would like to beat the metal as soon as possible. Any hints, pointers to TFM, will be sooo happily accepted. the oss emu10k1 driver has already good examples for dsp codes on sb live. but the interface is different between alsa and oss drivers, so you cannot use them straightforwardly. the oss emu10k1 archive also includes some documents about dsp codes on emu10k1. the problem is, as written above, the implementation of loader. how to manage static and dynamic routings, and so on. Exactly. OSS code does this management partly in the kernel. In my opinion, we should leave in kernel only necessary things, so the linker/loader/code manager should be written completely in the user space. This is basically what we have (the OSS guys ;). The information that is kept in the kernel is not meant to do any managing but instead it allows the patch loader/manager (implemented in user space) to retrieve the current DSP code state (which patches are loaded, which inputs connect to which outputs, ). This is very difficult (impossible, maybe) to deduce by simply retrieving the DSP code. Right, but we have things like shared memory for the inter-application communication. This job is not time-critical, so it would be really better to leave it outside the kernel space. And, at last, we can have more implementations of the linker/manager code, which is always appreciated in the open source community. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project http://www.alsa-project.org SuSE Linuxhttp://www.suse.com --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] rc2 stereo problem with via8233
At Thu, 27 Jun 2002 10:14:41 -0400, John covici wrote: OK, here is -xvv for that controller. 00:11.5 Multimedia audio controller: VIA Technologies, Inc. AC97 Audio Controller (rev 30) Subsystem: VIA Technologies, Inc.: Unknown device 4511 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin C routed to IRQ 11 Region 0: I/O ports at ec00 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 00: 06 11 59 30 01 00 10 02 30 00 01 04 00 00 00 00 10: 01 ec 00 00 00 00 00 00 00 00 00 00 00 00 00 00 20: 00 00 00 00 00 00 00 00 00 00 00 00 06 11 11 45 30: 00 00 00 00 c0 00 00 00 00 00 00 00 0b 03 00 00 and here is the -nvv 00:11.5 Class 0401: 1106:3059 (rev 30) Subsystem: 1106:4511 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin C routed to IRQ 11 Region 0: I/O ports at ec00 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- thanks. When I was using it, a stereo sample did play on both channels, from the cd, from pcm, they were mono -- I didn't test with stereo for the pcm. Is this sound card a multi channel and if so, how do I use more than one channel? i meant, to test wavefiles with mono samples and stereo samples, and check if any difference occurs. to be sure: the pcm with stereo samples comes out only from the left speaker, right? Takashi --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] 1869 trouble with snd-es18xx
Paul Davis wrote: Isn't there any option to use a 'virtual' ALSA device? Like snd-pcm-oss, but then the other way around. snd-oss-pcm module? Then you could use ALSA with a working OSS driver, and no working ALSA driver... ALSA has a significantly different internal architecture than OSS. It is not possible to use an OSS driver to support the ALSA midlevel code. What kind of midlevel code? the code in alsa-kernel/core/pcm (for the audio side). this is non-device-specific kernel side code. it is called by both user-space code via system calls and by device-specific code via direct function calls. I meant an ALSA driver, that you can assign an OSS device. The midlevel ALSA stuff can be handled by the ALSA driver, that just sends the audio data to the OSS device. you appear to be talking about routing audio data from alsa-lib to an actual OSS device, and also providing a translation of the entire alsa-lib audio API into OSS. it doesn't seem very likely that anyone from the ALSA would want to do this. the time could be much better spent on using an existing OSS driver to implement a native ALSA one. --p --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] alsa-api question
hi, i currently writing a plugin for alsa9 and i'm thinking of various paramters which could be set and if there are usefull for example does someone know 'good' values for snd_pcm_sw_params_set_xfer_align(), in the code samples it varies from 1 to 4, is it also usefull to set it to higher values for example 1024? and at least what is it good for? is there any need for any sw_parameters for 'normal' player applications?? thanx in advance joy --- Sponsored by: ThinkGeek at http://www.ThinkGeek.com/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] alsa-api question
i currently writing a plugin for alsa9 and i'm thinking of various
paramters which could be set and if there are usefull for example does
someone know 'good' values for snd_pcm_sw_params_set_xfer_align(), in the
code samples it varies from 1 to 4, is it also usefull to set it to higher
values for example 1024? and at least what is it good for?
is there any need for any sw_parameters for 'normal' player applications??
avail_min is the only sw_param you typically need to set, and even
that is only if you intend to use poll(2). all ALSA params come with
reasonable defaults. xfer_align, in particular, is one that i've
never ever bothered to set.
--p
\begin{ObJack}
ps. but why bother with all this when you can use JACK (http://jackit.sf.net/)?
\end{ObJack}
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Re: [Alsa-devel] rc2 stereo problem with via8233
At Thu, 27 Jun 2002 12:41:43 -0400, John covici wrote: OK, a stereo PCM does play both channels, a mono pcm plays left channel only. then try the following patch. Takashi via8233-fix.dif Description: Binary data
[Alsa-devel] AC97 ICH4 Public Documentation and Memeory mapped feature request
Many motherboards are coming out without the ICH2 compatability mode (IO access vs Memory Mapped). Here are links to the datasheets to add Memory Mapped functionality and other features of ICH4: General Data: http://www.intel.com/design/chipsets/845g/index.htm Datasheet: http://developer.intel.com/design/chipsets/datashts/290744.htm Chipset Comparisons: http://developer.intel.com/design/chipsets/linecard.htm?iid=ipp_browse+chpsts_compare Thanks, Gabe Widmer Software Developer SigmaTel Inc. [EMAIL PROTECTED]512-381-3746
[Alsa-devel] AC'97 HP out Register tied to Master out
I have to make a code change to the current released source to enable audio so it will work correctly with the common mixers in most distros. All AC'97 Windows drivers do this. I'll spit the code back once I am done. Many AC'97 audio designs use the HPout pin instead of the Master out. I also noticed that there is a switch to enable this through amixer. Anyway, the code I change will be delivered to my customers and I will feed the code back to you all. Thanks, Gabe Widmer Software Developer SigmaTel Inc. [EMAIL PROTECTED]512-381-3746
Re: [Alsa-devel] yes.. [was 1869 trouble with snd-es18xx
Hi, Finally I managed to get ALSA running properly on my Compaq Armada 3500 with an es1869 soundchip. The problem was apparently an IRQ DMA problem. I read the datasheet of the es1869 and there I found that it's best to have the first DMA channel set to 1 and the second DMA set to 0, 1 or 3. I haven't found anything about IRQ settings, but I tried irq 9, which didn't work, and now I'm using irq 5. However, Jose Andres Martinez runs the es18xx driver on his Armada 3500 with irq 9. The second DMA channel used to be 5. And the OSS sb driver worked perfectly, just like the Windows 98 and 2000 drivers. So that's pretty strange. On Windows I even managed to have full duplex. Haven't tried that yet. Maybe someone can add this information to the ALSA documentation somewhere. The ALSA HOWTO might be a good place? Jorn? greetz, Kasper P.S. me happy now :) finally I can use ALSA and JACK :) --- This sf.net email is sponsored by:ThinkGeek Bringing you mounds of caffeinated joy. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] yes.. [was 1869 trouble with snd-es18xx
On Fri, Jun 28, 2002 at 01:39:38AM +0200, Kasper Souren wrote: Finally I managed to get ALSA running properly on my Compaq Armada 3500 congratulations! Maybe someone can add this information to the ALSA documentation somewhere. The ALSA HOWTO might be a good place? Jorn? You can do it yourself: 1. Go to http://alsa.opensrc.org/index.php?page=es18xx 2. Click on Edit this document 3. Type the information and click save Regards, Mark --- This sf.net email is sponsored by:ThinkGeek Bringing you mounds of caffeinated joy. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Rate Conversion
Jaroslav Kysela wrote: On Thu, 27 Jun 2002, Kris Modrak wrote: I am writing a PCM application and wish to play a .wav file that has a sampling rate of 8kHz on a hardware setup that only supports sampling frequencies of 44.1 or 48kHz. Does anyone know how to implement this? You don't need to do this. Use the 'plughw' device which should do all conversions from you. I am a little confused about your advice. I am not sure how to access the 'plughw' device from my application. I have a full duplex application that can play a file at 44.1kHz and record at the same rate but I want it to operate at lower sampling frequencies. I am not sure of how I should be setting up my PCM devices. How do I tell them to convert, say, 8kHz audio data to 44.1kHz audio data so my hardware can play it at the correct speed? I tried using aplay with plughw to play an 8kHz file and got the following results aplay -Dplughw s1.wav Playing WAVE 's1.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Segmentation fault However it worked when using a 441.kHz file. Regards Kris Modrak --- This sf.net email is sponsored by:ThinkGeek Bringing you mounds of caffeinated joy. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
