Does LiveMedia take the conditions as follow into account:jitter caused by the network,skew due to tyhe asynchronous of two clock,and the adaption of the playout point in order to compensate the aforementioned conditions etc. We know, these are all very common in the network,a robust implementation should take them into account.Where do you achieve these in LiveMedia and how?
Our software, in accordance with the RTP/RTCP standard, delivers - to each client - data frames with an accurate presentation time. These presentation times are times in the *sender's* clock.
If the receiver's clock runs at a different rate than the sender's clock (the clock used by the data presentation times), then the receiving application may need to compensate for this. The mechanism that the application uses t odo so is (necessarily) application and codec-specific, and best handled by media decoders (which are not part of our software); therefore it is not something that we do in our libraries.
(For example, the receiving application might choose to drop some decoded audio samples, or to duplicate some decoded audio samples, as appropriate.)
-- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list [email protected] http://lists.live555.com/mailman/listinfo/live-devel
