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1 So, I'll describe how this stuff works.
3 The main modules:
5 1. stream.c: this is the input layer, this reads the input media (file, stdin,
6 vcd, dvd, network etc). what it has to know: appropriate buffering by
7 sector, seek, skip functions, reading by bytes, or blocks with any size.
8 The stream_t (stream.h) structure describes the input stream, file/device.
10 There is a stream cache layer (cache2.c), it's a wrapper for the stream
11 API. It does fork(), then emulates stream driver in the parent process,
12 and stream user in the child process, while proxying between them using
13 preallocated big memory chunk for FIFO buffer.
15 2. demuxer.c: this does the demultiplexing (separating) of the input to
16 audio, video or dvdsub channels, and their reading by buffered packages.
17 The demuxer.c is basically a framework, which is the same for all the
18 input formats, and there are parsers for each of them (mpeg-es,
19 mpeg-ps, avi, avi-ni, asf), these are in the demux_*.c files.
20 The structure is the demuxer_t. There is only one demuxer.
22 2.a. demux_packet_t, that is DP.
23 Contains one chunk (avi) or packet (asf,mpg). They are stored in memory as
24 in linked list, cause of their different size.
26 2.b. demuxer stream, that is DS.
27 Struct: demux_stream_t
28 Every channel (a/v/s) has one. This contains the packets for the stream
29 (see 2.a). For now, there can be 3 for each demuxer :
30 - audio (d_audio)
31 - video (d_video)
32 - DVD subtitle (d_dvdsub)
34 2.c. stream header. There are 2 types (for now): sh_audio_t and sh_video_t
35 This contains every parameter essential for decoding, such as input/output
36 buffers, chosen codec, fps, etc. There are each for every stream in
37 the file. At least one for video, if sound is present then another,
38 but if there are more, then there'll be one structure for each.
39 These are filled according to the header (avi/asf), or demux_mpg.c
40 does it (mpg) if it founds a new stream. If a new stream is found,
41 the ====> Found audio/video stream: <id> messages is displayed.
43 The chosen stream header and its demuxer are connected together
44 (ds->sh and sh->ds) to simplify the usage. So it's enough to pass the
45 ds or the sh, depending on the function.
47 For example: we have an asf file, 6 streams inside it, 1 audio, 5
48 video. During the reading of the header, 6 sh structs are created, 1
49 audio and 5 video. When it starts reading the packet, it chooses the
50 stream for the first found audio & video packet, and sets the sh
51 pointers of d_audio and d_video according to them. So later it reads
52 only these streams. Of course the user can force choosing a specific
53 stream with
54 -vid and -aid switches.
55 A good example for this is the DVD, where the english stream is not
56 always the first, so every VOB has different language :)
57 That's when we have to use for example the -aid 128 switch.
59 Now, how this reading works?
60 - demuxer.c/demux_read_data() is called, it gets how many bytes,
61 and where (memory address), would we like to read, and from which
62 DS. The codecs call this.
63 - this checks if the given DS's buffer contains something, if so, it
64 reads from there as much as needed. If there isn't enough, it calls
65 ds_fill_buffer(), which:
66 - checks if the given DS has buffered packages (DP's), if so, it moves
67 the oldest to the buffer, and reads on. If the list is empty, it
68 calls demux_fill_buffer() :
69 - this calls the parser for the input format, which reads the file
70 onward, and moves the found packages to their buffers.
71 Well it we'd like an audio package, but only a bunch of video
72 packages are available, then sooner or later the:
73 DEMUXER: Too many (%d in %d bytes) audio packets in the buffer
74 error shows up.
76 2.d. video.c: this file/function handle the reading and assembling of the
77 video frames. each call to video_read_frame() should read and return a
78 single video frame, and it's duration in seconds (float).
79 The implementation is splitted to 2 big parts - reading from mpeg-like
80 streams and reading from one-frame-per-chunk files (avi, asf, mov).
81 Then it calculates duration, either from fixed FPS value, or from the
82 PTS difference between and after reading the frame.
84 2.e. other utility functions: there are some useful code there, like
85 AVI muxer, or mp3 header parser, but leave them for now.
87 So everything is ok 'till now. It can be found in libmpdemux/ library.
88 It should compile outside of mplayer tree, you just have to implement few
89 simple functions, like mp_msg() to print messages, etc.
90 See libmpdemux/test.c for example.
92 See also formats.txt, for description of common media file formats and their
93 implementation details in libmpdemux.
95 Now, go on:
97 3. mplayer.c - ooh, he's the boss :)
98 Its main purpose is connecting the other modules, and maintaining A/V
99 sync.
101 The given stream's actual position is in the 'timer' field of the
102 corresponding stream header (sh_audio / sh_video).
104 The structure of the playing loop :
105 while(not EOF) {
106 fill audio buffer (read & decode audio) + increase a_frame
107 read & decode a single video frame + increase v_frame
108 sleep (wait until a_frame>=v_frame)
109 display the frame
110 apply A-V PTS correction to a_frame
111 handle events (keys,lirc etc) -> pause,seek,...
112 }
114 When playing (a/v), it increases the variables by the duration of the
115 played a/v.
116 - with audio this is played bytes / sh_audio->o_bps
117 Note: i_bps = number of compressed bytes for one second of audio
118 o_bps = number of uncompressed bytes for one second of audio
119 (this is = bps*samplerate*channels)
120 - with video this is usually == 1.0/fps, but I have to note that
121 fps doesn't really matters at video, for example asf doesn't have that,
122 instead there is "duration" and it can change per frame.
123 MPEG2 has "repeat_count" which delays the frame by 1-2.5 ...
124 Maybe only AVI and MPEG1 has fixed fps.
126 So everything works right until the audio and video are in perfect
127 synchronity, since the audio goes, it gives the timing, and if the
128 time of a frame passed, the next frame is displayed.
129 But what if these two aren't synchronized in the input file?
130 PTS correction kicks in. The input demuxers read the PTS (presentation
131 timestamp) of the packages, and with it we can see if the streams
132 are synchronized. Then MPlayer can correct the a_frame, within
133 a given maximal bounder (see -mc option). The summary of the
134 corrections can be found in c_total .
136 Of course this is not everything, several things suck.
137 For example the soundcards delay, which has to be corrected by
138 MPlayer! The audio delay is the sum of all these:
139 - bytes read since the last timestamp:
140 t1 = d_audio->pts_bytes/sh_audio->i_bps
141 - if Win32/ACM then the bytes stored in audio input buffer
142 t2 = a_in_buffer_len/sh_audio->i_bps
143 - uncompressed bytes in audio out buffer
144 t3 = a_buffer_len/sh_audio->o_bps
145 - not yet played bytes stored in the soundcard's (or DMA's) buffer
146 t4 = get_audio_delay()/sh_audio->o_bps
148 From this we can calculate what PTS we need for the just played
149 audio, then after we compare this with the video's PTS, we have
150 the difference!
152 Life didn't get simpler with AVI. There's the "official" timing
153 method, the BPS-based, so the header contains how many compressed
154 audio bytes or chunks belong to one second of frames.
155 In the AVI stream header there are 2 important fields, the
156 dwSampleSize, and dwRate/dwScale pairs:
157 - If the dwSampleSize is 0, then it's VBR stream, so its bitrate
158 isn't constant. It means that 1 chunk stores 1 sample, and
159 dwRate/dwScale gives the chunks/sec value.
160 - If the dwSampleSize is >0, then it's constant bitrate, and the
161 time can be measured this way: time = (bytepos/dwSampleSize) /
162 (dwRate/dwScale) (so the sample's number is divided with the
163 samplerate). Now the audio can be handled as a stream, which can
164 be cut to chunks, but can be one chunk also.
166 The other method can be used only for interleaved files: from
167 the order of the chunks, a timestamp (PTS) value can be calculated.
168 The PTS of the video chunks are simple: chunk number * fps
169 The audio is the same as the previous video chunk was.
170 We have to pay attention to the so called "audio preload", that is,
171 there is a delay between the audio and video streams. This is
172 usually 0.5-1.0 sec, but can be totally different.
173 The exact value was measured until now, but now the demux_avi.c
174 handles it: at the audio chunk after the first video, it calculates
175 the A/V difference, and take this as a measure for audio preload.
177 3.a. audio playback:
178 Some words on audio playback:
179 Not the playing is hard, but:
180 1. knowing when to write into the buffer, without blocking
181 2. knowing how much was played of what we wrote into
182 The first is needed for audio decoding, and to keep the buffer
183 full (so the audio will never skip). And the second is needed for
184 correct timing, because some soundcards delay even 3-7 seconds,
185 which can't be forgotten about.
186 To solve this, the OSS gives several possibilities:
187 - ioctl(SNDCTL_DSP_GETODELAY): tells how many unplayed bytes are in
188 the soundcard's buffer -> perfect for timing, but not all drivers
189 support it :(
190 - ioctl(SNDCTL_DSP_GETOSPACE): tells how much can we write into the
191 soundcard's buffer, without blocking. If the driver doesn't
192 support GETODELAY, we can use this to know how much the delay is.
193 - select(): should tell if we can write into the buffer without
194 blocking. Unfortunately it doesn't say how much we could :((
195 Also, doesn't/badly works with some drivers.
196 Only used if none of the above works.
198 4. Codecs. Consists of libmpcodecs/* and separate files or libs,
199 for example libmpeg2, loader, mp3lib.
201 mplayer.c doesn't call them directly, but through the dec_audio.c and
202 dec_video.c files, so the mplayer.c doesn't have to know anything about
203 the codecs.
205 libmpcodecs contains wrapper for every codecs, some of them include the
206 codec function implementation, some calls functions from other files
207 included with mplayer, some calls optional external libraries.
208 file naming convention in libmpcodecs:
209 ad_*.c - audio decoder (called through dec_audio.c)
210 vd_*.c - video decoder (called through dec_video.c)
211 ve_*.c - video encoder (used by mencoder)
212 vf_*.c - video filter (see option -vf)
214 On this topic, see also:
215 libmpcodecs.txt - The structure of the codec-filter path, with explanation
216 dr-methods.txt - Direct rendering, MPI buffer management for video codecs
217 codecs.conf.txt - How to write/edit codec configuration file (codecs.conf)
218 codec-devel.txt - Mike's hints about codec development - a bit OUTDATED
219 hwac3.txt - about SP/DIF audio passthrough
221 5. libvo: this displays the frame.
223 for details on this, read libvo.txt
225 6. libao2: this control audio playing
226 6.a audio plugins
228 for details on this, read libao2.txt