Max Kempter wrote: > hi, > > here is max again. > First of all I have to say that your answers are very helpfully, and week by week I`m nearer by my aim I want to do with PA. > > I have some trouble with: > ------------------------------------------------------------------------------------------------------------- > [/shell] > usr at CST-G2L2:~$ arecord | pacat -p -s 192.56.1.3 -d out -v > Opening a playback stream with sample specification 's16le 2ch 44100Hz'. > Connection established. > Stream successfully created. > Buffer metrics: maxlength=132300, tlength=88200, prebuf=87320, minreq=880 > Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono > Time: 0.000 sec; Latency: 1234567890ms (goes up to endless) > [shell/] > > The pacat-source is connected with the sink ?out?, but there is no level at volume meter at the reciver site. Each 30 s I see a short amplitude at the level meter, the amplitude goes up to 0dB and after 0.5 s back to circa ? 120dB. pacat and arecord have different defaults. You need to pass additional parameters to make sure the data is compatible. > > I reach the the host I want to reach but the packets are empty (or it send the wrong things). > > The second strange thing is that the Latency (I see it through -v option) goes up to endless, or till I push CTRL-C. > This shouldn't happen though. Can you check the latency on "out" when this happens? > If I try > [/shell] > arecord test.wav > [shell/] > It works and I record the mic capture. > > If I try > [/shell] > aplay test.wav | pacat -p -s 192.56.1.3 -d out > [shell/] > Also no problem, I recive the audio at alsa-sink ?out?. > Ehm... aplay doesn't send its stuff to stdout, so that is broken test case. Your "out" is probably your default alsa device. > ------------------------------------------------------ > > Let`s assume I load a source (eg mic) at Host1 and a load tunnel sink (eg out) from Host2, how can I route the source to the sink? > I want to do this without the modules module-rtp-{send/recv}! > > -------------------------------------------------------- I'm not familiar with module-rtp, so I can't tell you any more than what's in the docs. > > Next, every time I load a source/sink, with load-module, I have to read this message: > > [/shell] > >>> load-module module-alsa-source device=plughw:0,0 rate=48000 source_name=mic > > ALSA lib control.c:910:(snd_ctl_open_noupdate) Invalid CTL plughw:0,0 > alsa-util.c: Unable to attach to mixer plughw:0,0: File or Folder not found > [shell/] > Which mixer it means? Alsamixer? alsamixer is an application. What the message means is that it cannot find a mixer attached to that device. I don't think plug exports the mixer interface, only pcm. So you should use hw:0,0 as the device. > > My setup also works with this ?error? message, so it could be the best to ignore it? Or is it a problem with my kernel (2.6.16-2-686 stable)? > You just lose hw volume control. > ------------------------------------------------------ > > Which is the advantage of the module-null-sink? Is it for people who doesn`t have an audio device? > Amongst other things. It's also good for testing, and as a fallback when you have to rescue streams. > ------------------------------------------------------ > > And the last question is about the deamon parameter ?high-priority=1 what is the effect about this option? Is it the process priority at machine!? Yes. It will renice itself to -15 and (if possible) select FIFO scheduling. Rgds -- Pierre Ossman OpenSource-based Thin Client Technology System Developer Telephone: +46-13-21 46 00 Cendio AB Web: http://www.cendio.com -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 251 bytes Desc: OpenPGP digital signature URL: <http://lists.freedesktop.org/archives/pulseaudio-discuss/attachments/20061211/af54f9b4/attachment.pgp>