do you have any other sound hardware like a usb headset or anything you can compare it to. I really would like to know if that sound card sucks or something because I never see this in any of the stuff i am testing.
I can't even lab it up to try to fix. On Wed, May 7, 2008 at 9:46 AM, Csaba Zelei <[EMAIL PROTECTED]> wrote: > I tried it with the latest trunk. > If I set it to 60ms sometimes I still get <1ms rtp packet delta, if I set > it to 120ms then there is none > The rtp packet delta is still random within 50-70ms with sometimes too low > 15-30ms, sometimes too high 100-150ms delta (with codec-ms = 60ms), and with > 15-20ms jitter. > > > > Anthony Minessale wrote: > > Have you tried setting the codec-ms in the portaudio.conf.xml to 60 or 120 > ms? > Maybe the soundcard is not able to do 20ms intervals and portaudio is > doing the least common multiple and chopping it up for us. > I think what's happening is the timer in the module is set to the interval > from the config file (20ms) and during every 60ms period there is no audio > until the last ms. so in each 60 ms: > > 20ms (timeout..... flush buffer) > 20ms (timeout..... flush buffer) > 20ms (get 60ms worth of audio at once [3 20ms packets] but we have already > read 2 filler frames from the timeouts) > > So now we have read 5 packets instead of 3 and erased some of our buffer > because of perceived timeouts. > The code is using the assumption that if the device will obey the chosen > frame size and sample rate requests down to the interval. > > If you find and edit conf/autoload_configs/portaudio.conf.xml > > look for this: > > <param name="codec-ms" value="20"/> > > and change 20 to 60 > > Setting this to 60 will change the frame size of all the packets from 320 > to 960 and set the timer to clock at an interval of 60ms > Since the card seems to be able to reliably produce 3 20ms packets every > 60ms it should also be able to produce 1 60ms packet. > > FreeSWITCH should then buffer the audio and still deliver it over SIP at > 20ms if you want but you can opt to set the codec [EMAIL PROTECTED] to disable > buffering if you are in a reliable network. > > The same should be true for setting the codec-ms to 120 > > > On Wed, May 7, 2008 at 3:27 AM, Sluschny, Thomas < > [EMAIL PROTECTED]> wrote: > > > Hi Anthony, > > > > i also tested your patch with no success. > > As i already described below, the problem with all 60ms 3 packets comes > > from the soundcard. > > The hardware delivers its samples all 60 ms. > > Our problem is (like Csaba said) that we read out the buffer after 60ms, > > 3 times, each with samples for 20ms, AND WITH NO DELAY! > > So we get: 60ms wait and 3 RTP packets within <1ms to send, and after > > that we already wait 60 ms for the next samples. > > > > In my patch i wait appr.20 ms if last method call was no longer than 4ms > > ago, > > but i think we can do better with switch_core_timer_check() method, but > > i don't know exactly how. > > > > You are absolutly right with your demand for a better timing resolution > > under Windows, > > but this 60ms mystery is caused by the soundcard. > > > > Thomas > > > > ------------------------------ > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400
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