So, I'll describe how this stuff works.

The basis of the program's structure is basically logical, however it's
a big hack :)

The main modules:

1. streamer.c: this is the input, this reads the file or the VCD.
   what it has to know: appropriate buffering, seek, skip functions,
	 reading by bytes, or blocks with any size.
	 The stream_t structure describes the input stream, file/device.

2. demuxer.c: this does the demultiplexing of the input to audio and video
   channels, and their reading by buffered packages.
	 The demuxer.c is basically a framework, which is the same for all the
	 input formats, and there are parsers for each of them (mpeg-es,
	 mpeg-ps, avi, avi-ni, asf), these are in the demux_*.c files.
	 The structure is the demuxer_t. There is only one demuxer.

2.a. demuxer stream, that is DS. Its struct is demux_stream_t
   Every channel (a/v) has one.
	 For now, there can be 2 for each demuxer, one for the audio and one
	 for the video.

2.b. demux_packet_t, that is DP.
   This contains one chunk (avi) or packet (asf,mpg).
	 In the memory they are stored as chained lists, since they are of
	 different sizes.
	 
  Now, how this reading works?
	 - demuxer.c/demux_read_data() is called, it gets how many bytes,
	   and where (memory address), would we like to read, and from which
           DS. The codecs call this.
	 - this checks if the given DS's buffer contains something, if so, it
	   reads from there as much as needed. If there isn't enough, it calls
	   ds_fill_buffer(), which:
	 - checks if the given DS has buffered packages (DP's), if so, it moves
	   the oldest to the buffer, and reads on. If the list is empty, it
	   calls demux_fill_buffer() :
	 - this calls the parser for the input format, which reads the file
	   onward, and moves the found packages to their buffers.
		 Well it we'd like an audio package, but only a bunch of video
		 packages are available, then sooner or later the:
		 DEMUXER: Too many (%d in %d bytes) audio packets in the buffer
		 error shows up.

So everything is ok 'till now, I want to move them to a separate lib.

Now, go on:

3. mplayer.c - ooh, he's the boss :)
   The timing is solved odd, since it has/recommended to be done differently
	 for each of the formats, and sometimes can be done in many ways.
	 There are the a_frame and v_frame float variables, they store the
	 just played a/v position is seconds.
	 A new frame is displayed if v_frame<a_frame, and sound is decoded if
	 a_frame<v_frame.
	 When playing (a/v), it increases the variables by the duration of the
	 played a/v. In video, it's usually 1.0/fps, but I have to mention that
	 fps doesn't really matters at video, for example asf doesn't have that,
	 instead there is "duration" and it can change per frame.
	 MPEG2 has "repeat_count" which delays the frame by 1-2.5 ...
	 Maybe only AVI and MPEG1 has fixed fps.

	 So everything works right until the audio and video are in perfect
	 synchronity, since the audio goes, it gives the timing, and if the
	 time of a frame passed, the next frame is displayed.
	 But what if these two aren't synchronized in the input file?
	 PTS correction kicks in. The input demuxers read the PTS (presentation
	 timestamp) of the packages, and with it we can see if the streams
	 are synchronized. Then MPlayer can correct the a_frame, within
	 a given maximal bounder (see -mc option). The summary of the
	 corrections can be found in c_total .

	 Of course this is not everything, several things suck.
	 For example the soundcards delay, which has to be corrected by
	 MPlayer: that's why it needs the size of the audio buffer. It can
	 be measured with select(), which is unfortunately not supported by
	 every card... That's when it has to be given with the -abs option.
	 
	 Then there's another problem: in MPEG, the PTS is not given by
	 frames, rather by sectors, which can contain 10 frames, or only 0.1 .
	 In order this won't fuck up timing, we average the PTS by 5 frames,
	 and use this when correcting.
	 
	 Life didn't get simpler with AVI. There's the "official" timing
	 method, the BPS-based, so the header contains how many compressed
	 audio bytes belong to one second of frames.
	 Of course this doesn't always work... why it should :)
	 So I emulate the MPEG's PTS/sector method on AVI, that is the
	 AVI parser calculates a fake PTS for every read chunk, decided by
	 the type of the frames. This is how my timing is done. And sometimes
	 this works better.

	 In AVI, usually there is a bigger piece of audio stored first, then
	 comes the video. This needs to be calculated into the delay, this is
	 called "Initial PTS delay".
	 Of course there are 2 of them, one is stored in the header and not
	 really used :) the other isn't stored anywhere, this can only be
	 measured...
	 
4. Codecs. They are separate libs.
   For example libac3, libmpeg2, xa/*, alaw.c, opendivx/*, loader, mp3lib.
	 mplayer.c calls them if a piece of audio or video needs to be played.
	 (see the beginning of 3.)
	 And they call the appropriate demuxer, to get the compressed data.
	 (see 2.)

5. libvo: this displays the frame. There are 2 different output routines in it:

5.a draw_slice(): this displays YV12 pictures (3 frames, a full sized which
	 contains brightness, and 2 with 1/4 sizes, which contain the colour
	 info). MPEG codecs (libmpeg2, opendivx) use this. This doesn't have
	 to display the whole frame, only update small parts of it.
5.b draw_frame(): this is the older interface, this displays only complete
	 frames, and can do only packed format (YUY2, RGB/BGR).
	 Win32 codecs use this (DivX, Indeo, etc).