A NOTE has been added to this issue. ====================================================================== <https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2147> ====================================================================== Reported By: Gene Heskett Assigned To: tiwai ====================================================================== Project: ALSA - driver Issue ID: 2147 Category: PCI - atiixp Reproducibility: always Severity: major Priority: normal Status: assigned Distribution: FC5, uptodate Kernel Version: 2.6.16-1.2111_FC5 ====================================================================== Date Submitted: 05-23-2006 03:52 CEST Last Modified: 05-28-2006 03:08 CEST ====================================================================== Summary: lack of full duplex operation for ati-ixp chipset. Description: skype, ekiga, audacity, and generally any program that needs to both transmit and receive, using isolated paths, fails in playback when recording. In skype, I sound great at the far end, but reception may or may not work, with emphasis on the not. What I hear in the phones, or in the speakers, sounds like what you would get if you took a sentence of spoken words, break it up into say 5ms pieces, and play them at maybe 10 of these little snippets per second, so that the resultant sound is broken up to the point of no comprehension, and stretched in time by a considerable amount. In audacity, simultanious playback, or play one track while recording another, likewise fails in that only very short snippets of the played track are output to the phones or speakers. Likewise, ekiga suffers from this same effect, I'm told I sound great on teh other end, but I can't hear more than half a word occasionally. ====================================================================== ---------------------------------------------------------------------- tiwai - 05-23-06 19:14 ---------------------------------------------------------------------- Grr, you make the things too complex. Let's sort out the thing straight. The duplex problem and the inproper recording are different things. First, check whether both non-duplex playback and recording work properly. For checking playback, you can use speaker-test program in alsa-utils package, too. Run "speaker-test -c2 -tw", for example. As I mentioned, you can test recording of the played stream by setting "Capture Source" to "Mix". It needs no extra hardware setup, so no hard work after running a full marathon. If the full-duplex works by this way, skype and other programs should work as well -- as long as you set up the mixer correctly. Anyway, don't use audacity or whatever apps might be using OSS emulation for tests. They can't be used as references. Use basic tools included in ALSA packages for primary tests. ---------------------------------------------------------------------- Gene Heskett - 05-28-06 03:08 ---------------------------------------------------------------------- Ok, now I have a couple of days to play with this. To quote you: "Apparently you're trying to do something special." I suppose thats in the eye of the beholder, but I do not consider running skype or its ilk too terribly special. In my eye, what you are asking me to do is the special stuff. >From a few hours of screwing with this last week, its known that speaker-test runs ok regardless of the starting order. However if speaker-test is the first user of the audio system started, then nothing else can access in the headset now, so lets see if your example "arecord -vv -fcd foo.wav" works, and yes it does. However, no volume control appeared, and my talking, with the headset mouthpiece only half an inch to one side, looked as if it was overloading arecord. However, stopping the speaker-test, and "aplay foo.wav" also works. Now, start the recording first... And thats working also. So that, according to your judgement, means its working. So why can't I make it work in the real world? For skype or ekiga. FWIW, when I start arecord, I get this on the terminal screen but no volume control appears: [root@diablo libzrtp-0.2.0]# arecord -vv -fcd foo.wav Recording WAVE 'foo.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Warning: rate is not accurate (requested = 44100Hz, got = 48000Hz) please, try the plug plugin (-Dplug:default) Hardware PCM card 0 'ATI IXP' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 ##+ | 03%% Aborted by signal Interrupt... Does anyone have a recipe for an .asoundrc that will stop the above error, or make it work for skype etc? Audacity can record ok, PROVIDED it is the first thing started, if I do the speaker-test thing, then start audacity, then audacity is locked away from the device, this will also occur if I stop the audacity recording while speaker-test is running, and then attempt to restart a new recording. Is audacity an oss appliction? In which case what do we replace it with thats alsa compliant? -- Cheers, Gene Issue History Date Modified Username Field Change ====================================================================== 05-23-06 03:52 Gene Heskett New Issue 05-23-06 03:52 Gene Heskett Distribution => FC5, uptodate 05-23-06 03:52 Gene Heskett Kernel Version => 2.6.16-1.2111_FC5 05-23-06 15:01 tiwai Note Added: 0009934 05-23-06 16:46 Gene Heskett Note Added: 0009938 05-23-06 17:24 tiwai Note Added: 0009941 05-23-06 18:07 Gene Heskett Note Added: 0009943 05-23-06 19:14 tiwai Note Added: 0009947 05-28-06 03:08 Gene Heskett Note Added: 0009985 ====================================================================== _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-devel
