audio: add af_lavrresample, remove old resampling filters

[mplayer.git] / DOCS / tech / general.txt

So, I'll describe how this stuff works.

The main modules:

1. stream.c: this is the input layer, this reads the input media (file, stdin,
   vcd, dvd, network etc).  what it has to know: appropriate buffering by
   sector, seek, skip functions, reading by bytes, or blocks with any size.
   The stream_t (stream.h) structure describes the input stream, file/device.

   There is a stream cache layer (cache2.c), it's a wrapper for the stream
   API. It does fork(), then emulates stream driver in the parent process,
   and stream user in the child process, while proxying between them using
   preallocated big memory chunk for FIFO buffer.

2. demuxer.c: this does the demultiplexing (separating) of the input to
   audio, video or dvdsub 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. demux_packet_t, that is DP.
   Contains one chunk (avi) or packet (asf,mpg). They are stored in memory as
         in linked list, cause of their different size.

2.b. demuxer stream, that is DS.
   Struct: demux_stream_t
   Every channel (a/v/s) has one. This contains the packets for the stream
         (see 2.a). For now, there can be 3 for each demuxer :
         - audio (d_audio)
         - video (d_video)
         - DVD subtitle (d_dvdsub)

2.c. stream header. There are 2 types (for now): sh_audio_t and sh_video_t
   This contains every parameter essential for decoding, such as input/output
         buffers, chosen codec, fps, etc. There are each for every stream in
         the file. At least one for video, if sound is present then another,
         but if there are more, then there'll be one structure for each.
         These are filled according to the header (avi/asf), or demux_mpg.c
         does it (mpg) if it founds a new stream. If a new stream is found,
         the ====> Found audio/video stream: <id>  messages is displayed.

         The chosen stream header and its demuxer are connected together
         (ds->sh and sh->ds) to simplify the usage. So it's enough to pass the
         ds or the sh, depending on the function.

         For example: we have an asf file, 6 streams inside it, 1 audio, 5
         video. During the reading of the header, 6 sh structs are created, 1
         audio and 5 video. When it starts reading the packet, it chooses the
         stream for the first found audio & video packet, and sets the sh
         pointers of d_audio and d_video according to them. So later it reads
         only these streams. Of course the user can force choosing a specific
         stream with
         -vid and -aid switches.
         A good example for this is the DVD, where the english stream is not
         always the first, so every VOB has different language :)
         That's when we have to use for example the -aid 128 switch.

  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.

2.d. video.c: this file/function handle the reading and assembling of the
     video frames. each call to video_read_frame() should read and return a
     single video frame, and it's duration in seconds (float).
     The implementation is splitted to 2 big parts - reading from mpeg-like
     streams and reading from one-frame-per-chunk files (avi, asf, mov).
     Then it calculates duration, either from fixed FPS value, or from the
     PTS difference between and after reading the frame.

2.e. other utility functions: there are some useful code there, like
    AVI muxer, or mp3 header parser, but leave them for now.

So everything is ok 'till now. It can be found in libmpdemux/ library.
It should compile outside of mplayer tree, you just have to implement few
simple functions, like mp_msg() to print messages, etc.
See libmpdemux/test.c for example.

See also formats.txt, for description of common media file formats and their
implementation details in libmpdemux.

Now, go on:

3. mplayer.c - ooh, he's the boss :)
    Its main purpose is connecting the other modules, and maintaining A/V
    sync.

    The given stream's actual position is in the 'timer' field of the
    corresponding stream header (sh_audio / sh_video).

         The structure of the playing loop :
         while(not EOF) {
             fill audio buffer (read & decode audio) + increase a_frame
             read & decode a single video frame + increase v_frame
             sleep  (wait until a_frame>=v_frame)
             display the frame
             apply A-V PTS correction to a_frame
             handle events (keys,lirc etc) -> pause,seek,...
         }

         When playing (a/v), it increases the variables by the duration of the
         played a/v.
         - with audio this is played bytes / sh_audio->o_bps
         Note: i_bps = number of compressed bytes for one second of audio
               o_bps = number of uncompressed bytes for one second of audio
                   (this is = bps*samplerate*channels)
         - with video this is usually == 1.0/fps, but I have to note 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! The audio delay is the sum of all these:
         - bytes read since the last timestamp:
           t1 = d_audio->pts_bytes/sh_audio->i_bps
         - if Win32/ACM then the bytes stored in audio input buffer
           t2 = a_in_buffer_len/sh_audio->i_bps
         - uncompressed bytes in audio out buffer
           t3 = a_buffer_len/sh_audio->o_bps
         - not yet played bytes stored in the soundcard's (or DMA's) buffer
           t4 = get_audio_delay()/sh_audio->o_bps

         From this we can calculate what PTS we need for the just played
         audio, then after we compare this with the video's PTS, we have
         the difference!

         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 or chunks belong to one second of frames.
         In the AVI stream header there are 2 important fields, the
         dwSampleSize, and dwRate/dwScale pairs:
         - If the dwSampleSize is 0, then it's VBR stream, so its bitrate
         isn't constant. It means that 1 chunk stores 1 sample, and
         dwRate/dwScale gives the chunks/sec value.
         - If the dwSampleSize is >0, then it's constant bitrate, and the
         time can be measured this way:  time = (bytepos/dwSampleSize) /
         (dwRate/dwScale) (so the sample's number is divided with the
         samplerate). Now the audio can be handled as a stream, which can
         be cut to chunks, but can be one chunk also.

         The other method can be used only for interleaved files: from
         the order of the chunks, a timestamp (PTS) value can be calculated.
         The PTS of the video chunks are simple: chunk number * fps
         The audio is the same as the previous video chunk was.
         We have to pay attention to the so called "audio preload", that is,
         there is a delay between the audio and video streams. This is
         usually 0.5-1.0 sec, but can be totally different.
         The exact value was measured until now, but now the demux_avi.c
         handles it: at the audio chunk after the first video, it calculates
         the A/V difference, and take this as a measure for audio preload.

3.a. audio playback:
         Some words on audio playback:
         Not the playing is hard, but:
         1. knowing when to write into the buffer, without blocking
         2. knowing how much was played of what we wrote into
         The first is needed for audio decoding, and to keep the buffer
         full (so the audio will never skip). And the second is needed for
         correct timing, because some soundcards delay even 3-7 seconds,
         which can't be forgotten about.
         To solve this, the OSS gives several possibilities:
         - ioctl(SNDCTL_DSP_GETODELAY): tells how many unplayed bytes are in
           the soundcard's buffer -> perfect for timing, but not all drivers
           support it :(
         - ioctl(SNDCTL_DSP_GETOSPACE): tells how much can we write into the
           soundcard's buffer, without blocking. If the driver doesn't
           support GETODELAY, we can use this to know how much the delay is.
         - select(): should tell if we can write into the buffer without
           blocking. Unfortunately it doesn't say how much we could :((
           Also, doesn't/badly works with some drivers.
           Only used if none of the above works.

4. Codecs. Consists of libmpcodecs/* and separate files or libs,
   for example libmpeg2, loader, mp3lib.

   mplayer.c doesn't call them directly, but through the dec_audio.c and
   dec_video.c files, so the mplayer.c doesn't have to know anything about
   the codecs.

   libmpcodecs contains wrapper for every codecs, some of them include the
   codec function implementation, some calls functions from other files
   included with mplayer, some calls optional external libraries.
   file naming convention in libmpcodecs:
   ad_*.c - audio decoder (called through dec_audio.c)
   vd_*.c - video decoder (called through dec_video.c)
   ve_*.c - video encoder (used by mencoder)
   vf_*.c - video filter  (see option -vf)

   On this topic, see also:
   libmpcodecs.txt - The structure of the codec-filter path, with explanation
   dr-methods.txt - Direct rendering, MPI buffer management for video codecs
   codecs.conf.txt - How to write/edit codec configuration file (codecs.conf)
   codec-devel.txt - Mike's hints about codec development - a bit OUTDATED
   hwac3.txt - about SP/DIF audio passthrough

5. libvo: this displays the frame.

   for details on this, read libvo.txt

6. libao2: this control audio playing
6.a audio plugins

   for details on this, read libao2.txt