2001-03-18 13:57:06 +00:00
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So, I'll describe how this stuff works.
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The basis of the program's structure is basically logical, however it's
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a big hack :)
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The main modules:
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2001-04-20 20:48:25 +00:00
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1. streamer.c: this is the input, this reads the file or the VCD. what it has
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to know: appropriate buffering by sector, seek, skip functions,
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2001-03-18 13:57:06 +00:00
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reading by bytes, or blocks with any size.
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The stream_t structure describes the input stream, file/device.
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2001-03-18 15:48:47 +00:00
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2. demuxer.c: this does the demultiplexing of the input to audio and video
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2001-03-18 13:57:06 +00:00
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channels, and their reading by buffered packages.
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The demuxer.c is basically a framework, which is the same for all the
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input formats, and there are parsers for each of them (mpeg-es,
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mpeg-ps, avi, avi-ni, asf), these are in the demux_*.c files.
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The structure is the demuxer_t. There is only one demuxer.
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2001-04-20 20:48:25 +00:00
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2.a. demux_packet_t, that is DP.
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Contains one chunk (avi) or packet (asf,mpg). They are stored in memory as
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in chained list, cause of their different size.
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2.b. demuxer stream, that is DS. Struct: demux_stream_t
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Every channel (a/v) has one. This contains the packets for the stream (see
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2.a). For now, there can be 2 for each demuxer, one for the audio and
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one for the video.
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2.c. stream header. There are 2 types (for now): sh_audio_t and sh_video_t
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This contains every parameter essential for decoding, such as input/output
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buffers, chosen codec, fps, etc. There are each for every stream in
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the file. At least one for video, if sound is present then another,
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but if there are more, then there'll be one structure for each.
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These are filled according to the header (avi/asf), or demux_mpg.c
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does it (mpg) if it founds a new stream. If a new stream is found,
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the ====> Found audio/video stream: <id> messages is displayed.
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The chosen stream header and its demuxer are connected together
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(ds->sh and sh->ds) to simplify the usage. So it's enough to pass the
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ds or the sh, depending on the function.
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For example: we have an asf file, 6 streams inside it, 1 audio, 5
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video. During the reading of the header, 6 sh structs are created,
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1 audio and 5 video. When it starts reading the packet, it chooses
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the first found audio & video stream, and sets the sh pointers of
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d_audio and d_video according to them. So later it reads only these
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streams. Of course the user can force choosing a specific stream with
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-vid and -aid switches.
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A good example for this is the DVD, where the english stream is not
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always the first, so every VOB has different language :)
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That's when we have to use for example the -aid 128 switch.
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2001-03-18 13:57:06 +00:00
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Now, how this reading works?
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- demuxer.c/demux_read_data() is called, it gets how many bytes,
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and where (memory address), would we like to read, and from which
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DS. The codecs call this.
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- this checks if the given DS's buffer contains something, if so, it
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reads from there as much as needed. If there isn't enough, it calls
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ds_fill_buffer(), which:
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- checks if the given DS has buffered packages (DP's), if so, it moves
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the oldest to the buffer, and reads on. If the list is empty, it
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calls demux_fill_buffer() :
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- this calls the parser for the input format, which reads the file
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onward, and moves the found packages to their buffers.
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Well it we'd like an audio package, but only a bunch of video
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packages are available, then sooner or later the:
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DEMUXER: Too many (%d in %d bytes) audio packets in the buffer
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error shows up.
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So everything is ok 'till now, I want to move them to a separate lib.
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Now, go on:
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3. mplayer.c - ooh, he's the boss :)
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The timing is solved odd, since it has/recommended to be done differently
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2001-03-18 16:01:12 +00:00
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for each of the formats, and sometimes can be done in many ways.
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2001-03-18 13:57:06 +00:00
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There are the a_frame and v_frame float variables, they store the
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just played a/v position is seconds.
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A new frame is displayed if v_frame<a_frame, and sound is decoded if
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a_frame<v_frame.
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When playing (a/v), it increases the variables by the duration of the
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played a/v. In video, it's usually 1.0/fps, but I have to mention that
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2001-03-18 16:01:12 +00:00
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fps doesn't really matters at video, for example asf doesn't have that,
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instead there is "duration" and it can change per frame.
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2001-03-18 13:57:06 +00:00
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MPEG2 has "repeat_count" which delays the frame by 1-2.5 ...
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Maybe only AVI and MPEG1 has fixed fps.
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2001-03-18 16:01:12 +00:00
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So everything works right until the audio and video are in perfect
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2001-03-18 13:57:06 +00:00
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synchronity, since the audio goes, it gives the timing, and if the
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time of a frame passed, the next frame is displayed.
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But what if these two aren't synchronized in the input file?
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PTS correction kicks in. The input demuxers read the PTS (presentation
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timestamp) of the packages, and with it we can see if the streams
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are synchronized. Then MPlayer can correct the a_frame, within
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a given maximal bounder (see -mc option). The summary of the
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corrections can be found in c_total .
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Of course this is not everything, several things suck.
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For example the soundcards delay, which has to be corrected by
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MPlayer: that's why it needs the size of the audio buffer. It can
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be measured with select(), which is unfortunately not supported by
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every card... That's when it has to be given with the -abs option.
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Then there's another problem: in MPEG, the PTS is not given by
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frames, rather by sectors, which can contain 10 frames, or only 0.1 .
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In order this won't fuck up timing, we average the PTS by 5 frames,
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and use this when correcting.
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Life didn't get simpler with AVI. There's the "official" timing
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method, the BPS-based, so the header contains how many compressed
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audio bytes belong to one second of frames.
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Of course this doesn't always work... why it should :)
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So I emulate the MPEG's PTS/sector method on AVI, that is the
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AVI parser calculates a fake PTS for every read chunk, decided by
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the type of the frames. This is how my timing is done. And sometimes
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this works better.
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In AVI, usually there is a bigger piece of audio stored first, then
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comes the video. This needs to be calculated into the delay, this is
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called "Initial PTS delay".
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Of course there are 2 of them, one is stored in the header and not
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really used :) the other isn't stored anywhere, this can only be
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measured...
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4. Codecs. They are separate libs.
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For example libac3, libmpeg2, xa/*, alaw.c, opendivx/*, loader, mp3lib.
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mplayer.c calls them if a piece of audio or video needs to be played.
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(see the beginning of 3.)
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And they call the appropriate demuxer, to get the compressed data.
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(see 2.)
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2001-04-20 20:48:25 +00:00
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We have to pass the appropriate stream header as parameter (sh_audio/
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sh_video), this should contain all the needed info for decoding
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(the demuxer too: sh->ds).
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The codecs' seprating is underway, the audio is already done, the video is
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work-in-progress. The aim is that mplayer.c won't have to know
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which are the codecs and how to use 'em, instead it should call
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an init/decode audio/video function.
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5. libvo: this displays the frame.
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The constants for different pixelformats are defined in img_format.h,
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their usage is mandatory.
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Each vo driver _has_ to implement these:
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query_format() - queries if a given pixelformat is supported.
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return value: flags:
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0x1 - supported (by hardware or conversion)
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0x2 - supported (by hardware, without conversion)
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0x4 - sub/osd supported (has draw_alpha)
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IMPORTANT: it's mandatorial that every vo driver support the YV12 format,
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and one (or both) of BGR15 and BGR24, with conversion, if needed.
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If these aren't supported, not every codec will work! The mpeg codecs
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can output only YV12, and the older win32 DLLs only 15 and 24bpp.
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There is a fast MMX-using 15->16bpp converter, so it's not a
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significant speed-decrease!
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2001-03-18 13:57:06 +00:00
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2001-04-20 20:48:25 +00:00
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The BPP table, if the driver can't change bpp:
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current bpp has to accept these
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15 15
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16 15,16
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24 24
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24,32 24,32
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2001-03-18 13:57:06 +00:00
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2001-04-20 20:48:25 +00:00
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If it can change bpp (for example DGA 2, fbdev, svgalib), then if possible
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we have to change to the desired bpp. If the hardware doesn't support,
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we have to change to the one closest to it, and do conversion!
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init() - this is called before displaying of the first frame -
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initializing buffers, etc.
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draw_slice(): this displays YV12 pictures (3 planes, one full sized that
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contains brightness (Y), and 2 quarter-sized which the colour-info
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(U,V). MPEG codecs (libmpeg2, opendivx) use this. This doesn't have
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2001-03-18 13:57:06 +00:00
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to display the whole frame, only update small parts of it.
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2001-04-20 20:48:25 +00:00
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draw_frame(): this is the older interface, this displays only complete
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2001-03-18 14:27:28 +00:00
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frames, and can do only packed format (YUY2, RGB/BGR).
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2001-03-18 13:57:06 +00:00
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Win32 codecs use this (DivX, Indeo, etc).
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2001-04-20 20:48:25 +00:00
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draw_alpha(): this displays subtitles and OSD.
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It's a bit tricky to use it, since it's not a part of libvo API,
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but a callback-style stuff. The flip_page() has to call
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vo_draw_text(), so that it passes the size of the screen and the
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corresponding draw_alpha() implementation for the pixelformat
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(function pointer). The vo_draw_text() checks the characters to draw,
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and calls draw_alpha() for each.
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As a help, osd.c contains draw_alpha for each pixelformats, use this
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if possible!
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flip_page(): this is called after each frame, this diplays the buffer for
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real. This is 'swapbuffers' when double-buffering.
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