mirror of
https://github.com/mpv-player/mpv
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b0e0051746
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@1650 b3059339-0415-0410-9bf9-f77b7e298cf2
281 lines
13 KiB
Plaintext
281 lines
13 KiB
Plaintext
So, I'll describe how this stuff works.
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The main modules:
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1. streamer.c: this is the input layer, this reads the file or the VCD or
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stdin. what it has to know: appropriate buffering by sector, seek, skip
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functions, reading by bytes, or blocks with any size. The stream_t
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structure describes the input stream, file/device.
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2. demuxer.c: this does the demultiplexing of the input to audio and video
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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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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.
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Struct: demux_stream_t
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Every channel (a/v) has one. This contains the packets for the stream
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(see 2.a). For now, there can be 3 for each demuxer :
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- audio (d_audio)
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- video (d_video)
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- DVD subtitle (d_dvdsub)
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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, 1
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audio and 5 video. When it starts reading the packet, it chooses the
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stream for the first found audio & video packet, and sets the sh
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pointers of d_audio and d_video according to them. So later it reads
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only these streams. Of course the user can force choosing a specific
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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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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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Its main purpose is connecting the other modules, and maintaining A/V
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sync.
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The given stream's actual position is in the 'timer' field of the
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corresponding stream header (sh_audio / sh_video).
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The structure of the playing loop :
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while(not EOF) {
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fill audio buffer (read & decode audio) + increase a_frame
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read & decode a single video frame + increase v_frame
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sleep (wait until a_frame>=v_frame)
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display the frame
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apply A-V PTS correction to a_frame
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check for keys -> pause,seek,...
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}
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When playing (a/v), it increases the variables by the duration of the
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played a/v.
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- with audio this is played bytes / sh_audio->o_bps
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Note: i_bps = number of compressed bytes for one second of audio
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o_bps = number of uncompressed bytes for one second of audio
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(this is = bps*samplerate*channels)
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- with video this is usually == 1.0/fps, but I have to note that
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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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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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So everything works right until the audio and video are in perfect
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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! The audio delay is the sum of all these:
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- bytes read since the last timestamp:
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t1 = d_audio->pts_bytes/sh_audio->i_bps
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- if Win32/ACM then the bytes stored in audio input buffer
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t2 = a_in_buffer_len/sh_audio->i_bps
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- uncompressed bytes in audio out buffer
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t3 = a_buffer_len/sh_audio->o_bps
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- not yet played bytes stored in the soundcard's (or DMA's) buffer
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t4 = get_audio_delay()/sh_audio->o_bps
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From this we can calculate what PTS we need for the just played
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audio, then after we compare this with the video's PTS, we have
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the difference!
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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 or chunks belong to one second of frames.
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In the AVI stream header there are 2 important fields, the
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dwSampleSize, and dwRate/dwScale pairs:
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- If the dwSampleSize is 0, then it's VBR stream, so its bitrate
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isn't constant. It means that 1 chunk stores 1 sample, and
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dwRate/dwScale gives the chunks/sec value.
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- If the dwSampleSize is >0, then it's constant bitrate, and the
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time can be measured this way: time = (bytepos/dwSampleSize) /
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(dwRate/dwScale) (so the sample's number is divided with the
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samplerate). Now the audio can be handled as a stream, which can
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be cut to chunks, but can be one chunk also.
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The other method can be used only for interleaved files: from
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the order of the chunks, a timestamp (PTS) value can be calculated.
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The PTS of the video chunks are simple: chunk number * fps
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The audio is the same as the previous video chunk was.
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We have to pay attention to the so called "audio preload", that is,
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there is a delay between the audio and video streams. This is
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usually 0.5-1.0 sec, but can be totally different.
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The exact value was measured until now, but now the demux_avi.c
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handles it: at the audio chunk after the first video, it calculates
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the A/V difference, and take this as a measure for audio preload.
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3.a. audio playback:
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Some words on audio playback:
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Not the playing is hard, but:
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1. knowing when to write into the buffer, without blocking
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2. knowing how much was played of what we wrote into
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The first is needed for audio decoding, and to keep the buffer
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full (so the audio will never skip). And the second is needed for
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correct timing, because some soundcards delay even 3-7 seconds,
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which can't be forgotten about.
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To solve this, the OSS gives several possibilities:
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- ioctl(SNDCTL_DSP_GETODELAY): tells how many unplayed bytes are in
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the soundcard's buffer -> perfect for timing, but not all drivers
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support it :(
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- ioctl(SNDCTL_DSP_GETOSPACE): tells how much can we write into the
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soundcard's buffer, without blocking. If the driver doesn't
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support GETODELAY, we can use this to know how much the delay is.
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- select(): should tell if we can write into the buffer without
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blocking. Unfortunately it doesn't say how much we could :((
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Also, doesn't/badly works with some drivers.
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Only used if none of the above works.
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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 doesn't call the directly, but through the dec_audio.c and
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dec_video.c files, so the mplayer.c doesn't have to know anything about
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the codec.
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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-optimized 15->16bpp converter, so it's not a
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significant speed-decrease!
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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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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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to display the whole frame, only update small parts of it.
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draw_frame(): this is the older interface, this displays only complete
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frames, and can do only packed format (YUY2, RGB/BGR).
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Win32 codecs use this (DivX, Indeo, etc).
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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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6. libao2: this control audio playing
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As in libvo (see 5.) also here are some drivers, based on the same API:
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static int control(int cmd, int arg);
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This is for reading/setting driver-specific and other special parameters.
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Not really used for now.
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static int init(int rate,int channels,int format,int flags);
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The init of driver, opens device, sets sample rate, channels, sample format
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parameters.
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Sample format: usually AFMT_S16_LE or AFMT_U8, for more definitions see
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dec_audio.c and linux/soundcards.h files!
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static void uninit();
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Guess what.
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Ok I help: closes the device, not (yet) called when exit.
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static void reset();
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Resets device. To be exact, it's for deleting buffers' contents,
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so after reset() the previously received stuff won't be output.
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(called if pause or seek)
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static int get_space();
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Returns how many bytes can be written into the audio buffer without
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blocking (making caller process wait). If the buffer is (nearly) full,
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has to return 0!
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If it never gives 0, MPlayer won't work!
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static int play(void* data,int len,int flags);
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Plays a bit of audio, which is received throught the "data" memory area, with
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a size of "len". The "flags" isn't used yet. It has to copy the data, because
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they can be overwritten after the call is made. Doesn't really have to use
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all the bytes, it has to give back how many have been used (copied to
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buffer).
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static int get_delay();
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Has to return how many bytes are in the audio buffer. Be exact, if possible,
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since the whole timing depends on this! In the worst case, return the size
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of the buffer.
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!!! Because the video is synchronized to the audio (card), it's very important
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!!! that the get_space and get_delay functions be correctly implemented!
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