Re: [sipxtapi-dev] sipXtapi hardware requirements
On Wed, Apr 16, 2008 at 6:22 PM, Karsten Schubotz <
Karstenvip@xxxxxx> wrote:
Hi Keith,
Thank you for your quick and helpful answer,
regarding to the spezialized input and output audio drivers, do you have some more information? What kind of changes are necessary to get an efficient driver?
I would start with using a tool like 'aplay' and 'arec' -- ALSA tools that will play and record sound, and see if your driver can play and record 16bit signed little endian 8khz ulaw sound clearly (and wider bands, if you plan to support wideband, which I imagine you don't -- that support is very new).
You should be able to configure the size of the internal sound buffer the driver uses -- the smaller the buffer, the more on-time the CPU needs to be in serving the audio, and the lower latency the audio is -- which is needed for VoIP. sipXtapi in it's current form uses a 30ms audio driver buffer on windows, and I believe it is the same on Linux, or perhaps 20ms (I'm not currently looking at the source) using the new sipXtapi driver wrapper framework which is provided when you use the new flowgraph (see earlier entries in the mailing list for how to enable this). If you want to find out for yourself, you can do so by looking at the Mpid*.cpp|h and Mpod*.cpp|h files in sipXmediaLib.
You may also want to test using simple OSS play and record applications, as sipXtapi uses OSS in it's current incarnation. This will then test the impact that the OSS emulation layer has on the sound processing.
--
Keith Kyzivat
SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.comtel: +1 (617) 273-4000
