Rename DOCS/tech/ to DOCS/OUTATED-tech/

While DOCS/tech/ contains lots of documentation about mplayer's
internals, most of it seems outdated, and hasn't been touched in many
years. On the other hand, there still might be useful things in there,
but it's hard to tell which parts.

Instead of deleting all it, rename the directory to "warn" potential
developers that the documentation is completely outdated.

1 files changed, 228 insertions, 0 deletions

diff --git a/DOCS/OUTDATED-tech/general.txt b/DOCS/OUTDATED-tech/general.txt
new file mode 100644
index 0000000000..36a584b746
--- /dev/null
+++ b/DOCS/OUTDATED-tech/general.txt
@@ -0,0 +1,228 @@
+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