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These days we lunch a VOD (Video On Demand) project, since I am a fresh men to multimedia field and don't have any experience on audio, video, encoding and decoding issues. So I ask some simple questions, any proposal or advise is welcome. I want to discuss with anyone who is interested in this topic.
Now I have spend much time on researching some famous open source projects, such as VLC,Mpeg4IP, MPlayer. Since I try to use RTP as our file (audio and video) transfer protocol, so
the following items are the most important.
1. During server mode, how does the server packetize its audio and video data into RTP packets? And these are so many codec format, such as MPEG 1/2/4, ASF, XVID and so forth.
2. The second question is based on above, since when I meet a media file, how do I find its codec type and any other useful information from its file data? Known we can't just guess it from its backwared name, such *.rm,*.mp3.
3. In the client issue, when it receives data from remote area, encapsulated in RTP, of course we need to analysis its rtp header and get its payload data and then send to the decoder. Now my question is when packets lost, how do we deal with it? And buffer size, how much should we allocate in order not to be overflow?
It seems that firstly we should get audio and video data seperately from the corresponding media file, according to its code format.
Then we will map its audio or video data into rtp packets, since network transfer environment is error-prone we should take some action, such as fragment rules in order to make decoding can continue although packet lost.
Different codec have different rtp payload mapping format, and this is the key point we need to study further.