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Diffstat (limited to 'contrib/ffmpeg/doc')
| -rw-r--r-- | contrib/ffmpeg/doc/Makefile | 20 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/TODO | 82 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/faq.texi | 312 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/ffmpeg-doc.texi | 1607 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/ffmpeg_powerpc_performance_evaluation_howto.txt | 172 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/ffplay-doc.texi | 104 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/ffserver-doc.texi | 224 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/ffserver.conf | 349 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/hooks.texi | 113 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/optimization.txt | 158 | ||||
| -rw-r--r-- | contrib/ffmpeg/doc/soc.txt | 24 | ||||
| -rwxr-xr-x | contrib/ffmpeg/doc/texi2pod.pl | 427 |
12 files changed, 3592 insertions, 0 deletions
diff --git a/contrib/ffmpeg/doc/Makefile b/contrib/ffmpeg/doc/Makefile new file mode 100644 index 000000000..4fc9dfb8f --- /dev/null +++ b/contrib/ffmpeg/doc/Makefile @@ -0,0 +1,20 @@ +-include ../config.mak + +VPATH=$(SRC_PATH_BARE)/doc + +all: ffmpeg-doc.html faq.html ffserver-doc.html ffplay-doc.html hooks.html \ + ffmpeg.1 ffserver.1 ffplay.1 + +%.html: %.texi Makefile + texi2html -monolithic -number $< + +%.pod: %-doc.texi + ./texi2pod.pl $< $@ + +%.1: %.pod + pod2man --section=1 --center=" " --release=" " $< > $@ + +clean: + rm -f *.html *.pod *.1 + +.PHONY: all clean diff --git a/contrib/ffmpeg/doc/TODO b/contrib/ffmpeg/doc/TODO new file mode 100644 index 000000000..8271659d2 --- /dev/null +++ b/contrib/ffmpeg/doc/TODO @@ -0,0 +1,82 @@ +ffmpeg TODO list: +---------------- + +Fabrice's TODO list: (unordered) +------------------- +Short term: + +- seeking API and example in ffplay +- use AVFMTCTX_DISCARD_PKT in ffplay so that DV has a chance to work +- add RTSP regression test (both client and server) +- make ffserver allocate AVFormatContext +- clean up (incompatible change, for 0.5.0): + * AVStream -> AVComponent + * AVFormatContext -> AVInputStream/AVOutputStream + * suppress rate_emu from AVCodecContext +- add new float/integer audio filterting and conversion : suppress + CODEC_ID_PCM_xxc and use CODEC_ID_RAWAUDIO. +- fix telecine and frame rate conversion + +Long term (ask me if you want to help): + +- commit new imgconvert API and new PIX_FMT_xxx alpha formats +- commit new LGPL'ed float and integer-only AC3 decoder +- add WMA integer-only decoder +- add new MPEG4-AAC audio decoder (both integer-only and float version) + +Michael's TODO list: (unordered) (if anyone wanna help with sth, just ask) +------------------- +- optimize H264 CABAC +- more optimizations +- simper rate control + +Francois' TODO list: (unordered, without any timeframe) +------------------- +- test MACE decoder against the openquicktime one as suggested by A'rpi +- BeOS audio input grabbing backend +- BeOS video input grabbing backend +- have a REAL BeOS errno fix (return MKERROR(EXXX);), not a hack +- publish my BeOS libposix on BeBits so I can officially support ffserver :) +- check the whole code for thread-safety (global and init stuff) + +Philip'a TODO list: (alphabetically ordered) (please help) +------------------ +- Add a multi-ffm filetype so that feeds can be recorded into multiple files rather + than one big file. +- Authenticated users support -- where the authentication is in the URL +- Change ASF files so that the embedded timestamp in the frames is right rather + than being an offset from the start of the stream +- Make ffm files more resilient to changes in the codec structures so that you + can play old ffm files. + +unassigned TODO: (unordered) +--------------- +- use AVFrame for audio codecs too +- rework aviobuf.c buffering strategy and fix url_fskip +- generate optimal huffman tables for mjpeg encoding +- fix ffserver regression tests +- support xvids motion estimation +- support x264s motion estimation +- support x264s rate control +- SNOW: non translational motion compensation +- SNOW: more optimal quantization +- SNOW: 4x4 block support +- SNOW: 1/8 pel motion compensation support +- SNOW: iterative motion estimation based on subsampled images +- FLAC: lossy encoding (viterbi and naive scalar quantization) +- libavfilter +- JPEG2000 decoder & encoder +- MPEG4 GMC encoding support +- macroblock based pixel format (better cache locality, somewhat complex, one paper claimed it faster for high res) +- finish NUT implementation +- seeking regression test +- regression tests for codecs which dont have an encoder (I+P frame bitstream in svn) +- add support for using mplayers video filters to ffmpeg +- reverse engeneer RV30/RV40 +- finish implementation of WMV2 j-picture +- H264 encoder +- per MB ratecontrol (so VCD and such do work better) +- replace/rewrite libavcodec/fdctref.c +- write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions +- set up roundup bugtracker somewhere with (newBug, reproduced, analyzed, fixed, worksForMe, duplicate, wontFix, invalid, needMoreInfo, newPatch, ok, applied, rejected, needChanges, newRequest, implemented, wontImplement, invalidReq) states and a checked integer +- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc diff --git a/contrib/ffmpeg/doc/faq.texi b/contrib/ffmpeg/doc/faq.texi new file mode 100644 index 000000000..9f1e8ec2d --- /dev/null +++ b/contrib/ffmpeg/doc/faq.texi @@ -0,0 +1,312 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg FAQ +@titlepage +@sp 7 +@center @titlefont{FFmpeg FAQ} +@sp 3 +@end titlepage + + +@chapter General Problems + +@section I cannot read this file although this format seems to be supported by ffmpeg. + +Even if ffmpeg can read the file format, it may not support all its +codecs. Please consult the supported codec list in the ffmpeg +documentation. + +@section How do I encode JPEGs to another format ? + +If the JPEGs are named img1.jpg, img2.jpg, img3.jpg,..., use: + +@example + ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg +@end example + +@samp{%d} is replaced by the image number. + +@file{img%03d.jpg} generates @file{img001.jpg}, @file{img002.jpg}, etc... + +The same system is used for the other image formats. + +@section How do I encode movie to single pictures ? + +Use: + +@example + ffmpeg -i movie.mpg movie%d.jpg +@end example + +The @file{movie.mpg} used as input will be converted to +@file{movie1.jpg}, @file{movie2.jpg}, etc... + +Instead of relying on file format self-recognition, you may also use +@table @option +@item -vcodec ppm +@item -vcodec png +@item -vcodec mjpeg +@end table +to force the encoding. + +Applying that to the previous example: +@example + ffmpeg -i movie.mpg -f image2 -vcodec mjpeg menu%d.jpg +@end example + +Beware that there is no "jpeg" codec. Use "mjpeg" instead. + +@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it ? + +No. FFmpeg only supports open source codecs. Windows DLLs are not +portable, bloated and often slow. + +@section I get "Unsupported codec (id=86043) for input stream #0.1". What is the problem ? + +This is the Qcelp codec, FFmpeg has no support for that codec currently. Try mencoder/mplayer it might work. + +@section Why do I see a slight quality degradation with multithreaded MPEG* encoding ? + +For multithreaded MPEG* encoding, the encoded slices must be independent, +otherwise thread n would practically have to wait for n-1 to finish, so it's +quite logical that there is a small reduction of quality. This is not a bug. + +@section How can I read from the standard input or write to the standard output ? + +Use @file{-} as filename. + +@section Why does ffmpeg not decode audio in VOB files ? + +The audio is AC3 (a.k.a. A/52). AC3 decoding is an optional component in ffmpeg +as the component that handles AC3 decoding (liba52) is currently released under +the GPL. If you have liba52 installed on your system, enable AC3 decoding +with @code{./configure --enable-a52}. Take care: by +enabling AC3, you automatically change the license of libavcodec from +LGPL to GPL. + +@section Which codecs are supported by Windows ? + +Windows does not support standard formats like MPEG very well, unless you +install some additional codecs + +The following list of video codecs should work on most Windows systems: +@table @option +@item msmpeg4v2 +.avi/.asf +@item msmpeg4 +.asf only +@item wmv1 +.asf only +@item wmv2 +.asf only +@item mpeg4 +only if you have some MPEG-4 codec installed like ffdshow or XviD +@item mpeg1 +.mpg only +@end table +Note, ASF files often have .wmv or .wma extensions in Windows. It should also +be mentioned that Microsoft claims a patent on the ASF format, and may sue +or threaten users who create ASF files with non-Microsoft software. It is +strongly advised to avoid ASF where possible. + +The following list of audio codecs should work on most Windows systems: +@table @option +@item adpcm_ima_wav +@item adpcm_ms +@item pcm +@item mp3 +if some MP3 codec like LAME is installed +@end table + +@section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux? + +This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at +(@url{http://mplayerhq.hu/~michael/bttv-420-2.4.26.patch}). This may also +apply cleanly to other 2.4-series kernels. + +@section How do I avoid the ugly aliasing artifacts in bt8x8 captures on Linux? + +Pass 'combfilter=1 lumafilter=1' to the bttv driver. Note though that 'combfilter=1' +will cause somewhat too strong filtering. A fix is to apply (@url{http://mplayerhq.hu/~michael/bttv-comb-2.4.26.patch}) +or (@url{http://mplayerhq.hu/~michael/bttv-comb-2.6.6.patch}) +and pass 'combfilter=2'. + +@section I have a problem with an old version of ffmpeg; where should I report it? +Nowhere. Upgrade to the latest release or if there is no recent release upgrade +to Subversion HEAD. You could also try to report it. Maybe you will get lucky and +become the first person in history to get an answer different from "upgrade +to Subversion HEAD". + +@section -f jpeg doesn't work. + +Try '-f image2 test%d.jpg'. + +@section Why can I not change the framerate? + +Some codecs, like MPEG-1/2, only allow a small number of fixed framerates. +Choose a different codec with the -vcodec command line option. + +@section ffmpeg does not work; What is wrong? + +Try a 'make distclean' in the ffmpeg source directory. If this does not help see +(@url{http://ffmpeg.org/bugreports.php}). + +@section How do I encode XviD or DivX video with ffmpeg? + +Both XviD and DivX (version 4+) are implementations of the ISO MPEG-4 +standard (note that there are many other coding formats that use this +same standard). Thus, use '-vcodec mpeg4' to encode these formats. The +default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want +a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will +force the fourcc 'xvid' to be stored as the video fourcc rather than the +default. + +@section How do I encode videos which play on the iPod? + +@table @option +@item needed stuff +-acodec aac -vcodec mpeg4 width<=320 height<=240 +@item working stuff +4mv, title +@item non-working stuff +B-frames +@item example command line +ffmpeg -i input -acodec aac -ab 128 -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv+trell -aic 2 -cmp 2 -subcmp 2 -s 320x180 -title X output.mp4 +@end table + +@section How do I encode videos which play on the PSP? + +@table @option +@item needed stuff +-acodec aac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp +@item working stuff +4mv, title +@item non-working stuff +B-frames +@item example command line +ffmpeg -i input -acodec aac -ab 128 -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv+trell -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp output.mp4 +@item needed stuff for H.264 +-acodec aac -vcodec h264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp +@item working stuff for H.264 +title, loop filter +@item non-working stuff for H.264 +CAVLC +@item example command line +ffmpeg -i input -acodec aac -ab 128 -vcodec h264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4 +@end table + +@section How can I read DirectShow files? + +If you have built FFmpeg with @code{./configure --enable-avisynth} +(only possible on MinGW/Cygwin platforms), +then you may use any file that DirectShow can read as input. +(Be aware that this feature has been recently added, +so you will need to help yourself in case of problems.) + +Just create an "input.avs" text file with this single line ... +@example + DirectShowSource("C:\path to your file\yourfile.asf") +@end example +... and then feed that text file to FFmpeg: +@example + ffmpeg -i input.avs +@end example + +For ANY other help on Avisynth, please visit @url{http://www.avisynth.org/}. + +@chapter Development + +@section When will the next FFmpeg version be released? / Why are FFmpeg releases so few and far between? + +Like most open source projects FFmpeg suffers from a certain lack of +manpower. For this reason the developers have to prioritize the work +they do and putting out releases is not at the top of the list, fixing +bugs and reviewing patches takes precedence. Please don't complain or +request more timely and/or frequent releases unless you are willing to +help out creating them. + +@section Why doesn't FFmpeg support feature [xyz]? + +Because no one has taken on that task yet. FFmpeg development is +driven by the tasks that are important to the individual developers. +If there is a feature that is important to you, the best way to get +it implemented is to undertake the task yourself. + + +@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat ? + +Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively, +examine the source code for one of the many open source projects that +already incorporate ffmpeg at (@url{projects.php}). + +@section Can you support my C compiler XXX ? + +No. Only GCC is supported. GCC is ported to most systems available and there +is no need to pollute the source code with @code{#ifdef}s +related to the compiler. + +@section Can I use FFmpeg or libavcodec under Windows ? + +Yes, but the MinGW tools @emph{must} be used to compile FFmpeg. You +can link the resulting DLLs with any other Windows program. Read the +@emph{Native Windows Compilation} and @emph{Visual C++ compatibility} +sections in the FFmpeg documentation to find more information. + +@section Can you add automake, libtool or autoconf support ? + +No. These tools are too bloated and they complicate the build. Moreover, +since only @samp{gcc} is supported they would add little advantages in +terms of portability. + +@section Why not rewrite ffmpeg in object-oriented C++ ? + +ffmpeg is already organized in a highly modular manner and does not need to +be rewritten in a formal object language. Further, many of the developers +favor straight C; it works for them. For more arguments on this matter, +read "Programming Religion" at (@url{http://lkml.org/faq/lkmlfaq-15.html}). + +@section Why are the ffmpeg programs devoid of debugging symbols ? + +The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug +information. Those binaries are strip'd to create ffmpeg, ffplay, etc. If +you need the debug information, used the *_g versions. + +@section I do not like the LGPL, can I contribute code under the GPL instead ? + +Yes, as long as the code is optional and can easily and cleanly be placed +under #ifdef CONFIG_GPL without breaking anything. So for example a new codec +or filter would be OK under GPL while a bugfix to LGPL code would not. + +@section I want to compile xyz.c alone but my compiler produced many errors. + +Common code is in its own files in libav* and is used by the individual +codecs. They will not work without the common parts, you have to compile +the whole libav*. If you wish, disable some parts with configure switches. +You can also try to hack it and remove more, but if you had problems fixing +the compilation failure then you are probably not qualified for this. + +@section Visual C++ produces many errors. + +Visual C++ is not compliant to the C standard and does not support +the inline assembly used in FFmpeg. +If you wish - for whatever weird reason - to use Visual C++ for your +project then you can link the Visual C++ code with libav* as long as +you compile the latter with a working C compiler. For more information, see +the @emph{Visual C++ compatibility} section in the FFmpeg documentation. + +There have been efforts to make FFmpeg compatible with Visual C++ in the +past. However, they have all been rejected as too intrusive, especially +since MinGW does the job perfectly adequately. None of the core developers +work with Visual C++ and thus this item is low priority. Should you find +the silver bullet that solves this problem, feel free to shoot it at us. + +@section I have a file in memory / a API different from *open/*read/ libc how do i use it with libavformat ? + +You have to implement a URLProtocol, see libavformat/file.c in FFmpeg +and libmpdemux/demux_lavf.c in MPlayer sources. + +@section I get "No compatible shell script interpreter found." in MSys. + +The standard MSys bash (2.04) is broken. You need to install 2.05 or later. + +@bye diff --git a/contrib/ffmpeg/doc/ffmpeg-doc.texi b/contrib/ffmpeg/doc/ffmpeg-doc.texi new file mode 100644 index 000000000..2d814c0fb --- /dev/null +++ b/contrib/ffmpeg/doc/ffmpeg-doc.texi @@ -0,0 +1,1607 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Documentation +@titlepage +@sp 7 +@center @titlefont{FFmpeg Documentation} +@sp 3 +@end titlepage + + +@chapter Introduction + +FFmpeg is a very fast video and audio converter. It can also grab from +a live audio/video source. + +The command line interface is designed to be intuitive, in the sense +that FFmpeg tries to figure out all parameters that can possibly be +derived automatically. You usually only have to specify the target +bitrate you want. + +FFmpeg can also convert from any sample rate to any other, and resize +video on the fly with a high quality polyphase filter. + +@chapter Quick Start + +@c man begin EXAMPLES +@section Video and Audio grabbing + +FFmpeg can use a video4linux compatible video source and any Open Sound +System audio source: + +@example +ffmpeg /tmp/out.mpg +@end example + +Note that you must activate the right video source and channel before +launching FFmpeg with any TV viewer such as xawtv +(@url{http://bytesex.org/xawtv/}) by Gerd Knorr. You also +have to set the audio recording levels correctly with a +standard mixer. + +@section Video and Audio file format conversion + +* FFmpeg can use any supported file format and protocol as input: + +Examples: + +* You can use YUV files as input: + +@example +ffmpeg -i /tmp/test%d.Y /tmp/out.mpg +@end example + +It will use the files: +@example +/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, +/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc... +@end example + +The Y files use twice the resolution of the U and V files. They are +raw files, without header. They can be generated by all decent video +decoders. You must specify the size of the image with the @option{-s} option +if FFmpeg cannot guess it. + +* You can input from a raw YUV420P file: + +@example +ffmpeg -i /tmp/test.yuv /tmp/out.avi +@end example + +test.yuv is a file containing raw YUV planar data. Each frame is composed +of the Y plane followed by the U and V planes at half vertical and +horizontal resolution. + +* You can output to a raw YUV420P file: + +@example +ffmpeg -i mydivx.avi hugefile.yuv +@end example + +* You can set several input files and output files: + +@example +ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg +@end example + +Converts the audio file a.wav and the raw YUV video file a.yuv +to MPEG file a.mpg. + +* You can also do audio and video conversions at the same time: + +@example +ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2 +@end example + +Converts a.wav to MPEG audio at 22050Hz sample rate. + +* You can encode to several formats at the same time and define a +mapping from input stream to output streams: + +@example +ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0 +@end example + +Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map +file:index' specifies which input stream is used for each output +stream, in the order of the definition of output streams. + +* You can transcode decrypted VOBs + +@example +ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec mp3 -ab 128 snatch.avi +@end example + +This is a typical DVD ripping example; the input is a VOB file, the +output an AVI file with MPEG-4 video and MP3 audio. Note that in this +command we use B-frames so the MPEG-4 stream is DivX5 compatible, and +GOP size is 300 which means one intra frame every 10 seconds for 29.97fps +input video. Furthermore, the audio stream is MP3-encoded so you need +to enable LAME support by passing @code{--enable-mp3lame} to configure. +The mapping is particularly useful for DVD transcoding +to get the desired audio language. + +NOTE: To see the supported input formats, use @code{ffmpeg -formats}. +@c man end + +@chapter Invocation + +@section Syntax + +The generic syntax is: + +@example +@c man begin SYNOPSIS +ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}... +@c man end +@end example +@c man begin DESCRIPTION +If no input file is given, audio/video grabbing is done. + +As a general rule, options are applied to the next specified +file. Therefore, order is important, and you can have the same +option on the command line multiple times. Each occurrence is +then applied to the next input or output file. + +* To set the video bitrate of the output file to 64kbit/s: +@example +ffmpeg -i input.avi -b 64k output.avi +@end example + +* To force the frame rate of the input and output file to 24 fps: +@example +ffmpeg -r 24 -i input.avi output.avi +@end example + +* To force the frame rate of the output file to 24 fps: +@example +ffmpeg -i input.avi -r 24 output.avi +@end example + +* To force the frame rate of input file to 1 fps and the output file to 24 fps: +@example +ffmpeg -r 1 -i input.avi -r 24 output.avi +@end example + +The format option may be needed for raw input files. + +By default, FFmpeg tries to convert as losslessly as possible: It +uses the same audio and video parameters for the outputs as the one +specified for the inputs. +@c man end + +@c man begin OPTIONS +@section Main options + +@table @option +@item -L +Show license. + +@item -h +Show help. + +@item -version +Show version. + +@item -formats +Show available formats, codecs, protocols, ... + +@item -f fmt +Force format. + +@item -i filename +input filename + +@item -y +Overwrite output files. + +@item -t duration +Set the recording time in seconds. +@code{hh:mm:ss[.xxx]} syntax is also supported. + +@item -fs limit_size +Set the file size limit. + +@item -ss position +Seek to given time position in seconds. +@code{hh:mm:ss[.xxx]} syntax is also supported. + +@item -itsoffset offset +Set the input time offset in seconds. +@code{[-]hh:mm:ss[.xxx]} syntax is also supported. +This option affects all the input files that follow it. +The offset is added to the timestamps of the input files. +Specifying a positive offset means that the corresponding +streams are delayed by 'offset' seconds. + +@item -title string +Set the title. + +@item -timestamp time +Set the timestamp. + +@item -author string +Set the author. + +@item -copyright string +Set the copyright. + +@item -comment string +Set the comment. + +@item -album string +Set the album. + +@item -track number +Set the track. + +@item -year number +Set the year. + +@item -v verbose +Control amount of logging. + +@item -target type +Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd", +"ntsc-svcd", ... ). All the format options (bitrate, codecs, +buffer sizes) are then set automatically. You can just type: + +@example +ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg +@end example + +Nevertheless you can specify additional options as long as you know +they do not conflict with the standard, as in: + +@example +ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg +@end example + +@item -dframes number +Set the number of data frames to record. + +@item -scodec codec +Force subtitle codec ('copy' to copy stream). + +@item -newsubtitle +Add a new subtitle stream to the current output stream. + +@item -slang code +Set the ISO 639 language code (3 letters) of the current subtitle stream. + +@end table + +@section Video Options + +@table @option +@item -b bitrate +Set the video bitrate in bit/s (default = 200 kb/s). +@item -vframes number +Set the number of video frames to record. +@item -r fps +Set frame rate (Hz value, fraction or abbreviation), (default = 25). +@item -s size +Set frame size. The format is @samp{wxh} (default = 160x128). +The following abbreviations are recognized: +@table @samp +@item sqcif +128x96 +@item qcif +176x144 +@item cif +352x288 +@item 4cif +704x576 +@end table + +@item -aspect aspect +Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777). +@item -croptop size +Set top crop band size (in pixels). +@item -cropbottom size +Set bottom crop band size (in pixels). +@item -cropleft size +Set left crop band size (in pixels). +@item -cropright size +Set right crop band size (in pixels). +@item -padtop size +Set top pad band size (in pixels). +@item -padbottom size +Set bottom pad band size (in pixels). +@item -padleft size +Set left pad band size (in pixels). +@item -padright size +Set right pad band size (in pixels). +@item -padcolor (hex color) +Set color of padded bands. The value for padcolor is expressed +as a six digit hexadecimal number where the first two digits +represent red, the middle two digits green and last two digits +blue (default = 000000 (black)). +@item -vn +Disable video recording. +@item -bt tolerance +Set video bitrate tolerance (in bit/s). +@item -maxrate bitrate +Set max video bitrate tolerance (in bit/s). +@item -minrate bitrate +Set min video bitrate tolerance (in bit/s). +@item -bufsize size +Set rate control buffer size (in bits). +@item -vcodec codec +Force video codec to @var{codec}. Use the @code{copy} special value to +tell that the raw codec data must be copied as is. +@item -sameq +Use same video quality as source (implies VBR). + +@item -pass n +Select the pass number (1 or 2). It is useful to do two pass +encoding. The statistics of the video are recorded in the first +pass and the video is generated at the exact requested bitrate +in the second pass. + +@item -passlogfile file +Set two pass logfile name to @var{file}. + +@item -newvideo +Add a new video stream to the current output stream. + +@end table + +@section Advanced Video Options + +@table @option +@item -pix_fmt format +Set pixel format. +@item -g gop_size +Set the group of pictures size. +@item -intra +Use only intra frames. +@item -vdt n +Discard threshold. +@item -qscale q +Use fixed video quantizer scale (VBR). +@item -qmin q +minimum video quantizer scale (VBR) +@item -qmax q +maximum video quantizer scale (VBR) +@item -qdiff q +maximum difference between the quantizer scales (VBR) +@item -qblur blur +video quantizer scale blur (VBR) +@item -qcomp compression +video quantizer scale compression (VBR) + +@item -lmin lambda +minimum video lagrange factor (VBR) +@item -lmax lambda +max video lagrange factor (VBR) +@item -mblmin lambda +minimum macroblock quantizer scale (VBR) +@item -mblmax lambda +maximum macroblock quantizer scale (VBR) + +These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units, +but you may use the QP2LAMBDA constant to easily convert from 'q' units: +@example +ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext +@end example + +@item -rc_init_cplx complexity +initial complexity for single pass encoding +@item -b_qfactor factor +qp factor between P- and B-frames +@item -i_qfactor factor +qp factor between P- and I-frames +@item -b_qoffset offset +qp offset between P- and B-frames +@item -i_qoffset offset +qp offset between P- and I-frames +@item -rc_eq equation +Set rate control equation (@pxref{FFmpeg formula +evaluator}) (default = @code{tex^qComp}). +@item -rc_override override +rate control override for specific intervals +@item -me method +Set motion estimation method to @var{method}. +Available methods are (from lowest to best quality): +@table @samp +@item zero +Try just the (0, 0) vector. +@item phods +@item log +@item x1 +@item epzs +(default method) +@item full +exhaustive search (slow and marginally better than epzs) +@end table + +@item -dct_algo algo +Set DCT algorithm to @var{algo}. Available values are: +@table @samp +@item 0 +FF_DCT_AUTO (default) +@item 1 +FF_DCT_FASTINT +@item 2 +FF_DCT_INT +@item 3 +FF_DCT_MMX +@item 4 +FF_DCT_MLIB +@item 5 +FF_DCT_ALTIVEC +@end table + +@item -idct_algo algo +Set IDCT algorithm to @var{algo}. Available values are: +@table @samp +@item 0 +FF_IDCT_AUTO (default) +@item 1 +FF_IDCT_INT +@item 2 +FF_IDCT_SIMPLE +@item 3 +FF_IDCT_SIMPLEMMX +@item 4 +FF_IDCT_LIBMPEG2MMX +@item 5 +FF_IDCT_PS2 +@item 6 +FF_IDCT_MLIB +@item 7 +FF_IDCT_ARM +@item 8 +FF_IDCT_ALTIVEC +@item 9 +FF_IDCT_SH4 +@item 10 +FF_IDCT_SIMPLEARM +@end table + +@item -er n +Set error resilience to @var{n}. +@table @samp +@item 1 +FF_ER_CAREFUL (default) +@item 2 +FF_ER_COMPLIANT +@item 3 +FF_ER_AGGRESSIVE +@item 4 +FF_ER_VERY_AGGRESSIVE +@end table + +@item -ec bit_mask +Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of +the following values: +@table @samp +@item 1 +FF_EC_GUESS_MVS (default = enabled) +@item 2 +FF_EC_DEBLOCK (default = enabled) +@end table + +@item -bf frames +Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4). +@item -mbd mode +macroblock decision +@table @samp +@item 0 +FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg). +@item 1 +FF_MB_DECISION_BITS: Choose the one which needs the fewest bits. +@item 2 +FF_MB_DECISION_RD: rate distortion +@end table + +@item -4mv +Use four motion vector by macroblock (MPEG-4 only). +@item -part +Use data partitioning (MPEG-4 only). +@item -bug param +Work around encoder bugs that are not auto-detected. +@item -strict strictness +How strictly to follow the standards. +@item -aic +Enable Advanced intra coding (h263+). +@item -umv +Enable Unlimited Motion Vector (h263+) + +@item -deinterlace +Deinterlace pictures. +@item -ilme +Force interlacing support in encoder (MPEG-2 and MPEG-4 only). +Use this option if your input file is interlaced and you want +to keep the interlaced format for minimum losses. +The alternative is to deinterlace the input stream with +@option{-deinterlace}, but deinterlacing introduces losses. +@item -psnr +Calculate PSNR of compressed frames. +@item -vstats +Dump video coding statistics to @file{vstats_HHMMSS.log}. +@item -vhook module +Insert video processing @var{module}. @var{module} contains the module +name and its parameters separated by spaces. +@item -top n +top=1/bottom=0/auto=-1 field first +@item -dc precision +Intra_dc_precision. +@item -vtag fourcc/tag +Force video tag/fourcc. +@item -qphist +Show QP histogram. +@item -vbsf bitstream filter +Bitstream filters available are "dump_extra", "remove_extra", "noise". +@end table + +@section Audio Options + +@table @option +@item -aframes number +Set the number of audio frames to record. +@item -ar freq +Set the audio sampling frequency (default = 44100 Hz). +@item -ab bitrate +Set the audio bitrate in kbit/s (default = 64). +@item -ac channels +Set the number of audio channels (default = 1). +@item -an +Disable audio recording. +@item -acodec codec +Force audio codec to @var{codec}. Use the @code{copy} special value to +specify that the raw codec data must be copied as is. +@item -newaudio +Add a new audio track to the output file. If you want to specify parameters, +do so before @code{-newaudio} (@code{-acodec}, @code{-ab}, etc..). + +Mapping will be done automatically, if the number of output streams is equal to +the number of input streams, else it will pick the first one that matches. You +can override the mapping using @code{-map} as usual. + +Example: +@example +ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384 test.mpg -acodec mp2 -ab 192 -newaudio +@end example +@item -alang code +Set the ISO 639 language code (3 letters) of the current audio stream. +@end table + +@section Advanced Audio options: + +@table @option +@item -atag fourcc/tag +Force audio tag/fourcc. +@item -absf bitstream filter +Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp". +@end table + +@section Subtitle options: + +@table @option +@item -scodec codec +Force subtitle codec ('copy' to copy stream). +@item -newsubtitle +Add a new subtitle stream to the current output stream. +@item -slang code +Set the ISO 639 language code (3 letters) of the current subtitle stream. +@end table + +@section Audio/Video grab options + +@table @option +@item -vd device +sEt video grab device (e.g. @file{/dev/video0}). +@item -vc channel +Set video grab channel (DV1394 only). +@item -tvstd standard +Set television standard (NTSC, PAL (SECAM)). +@item -dv1394 +Set DV1394 grab. +@item -ad device +Set audio device (e.g. @file{/dev/dsp}). +@item -grab format +Request grabbing using. +@item -gd device +Set grab device. +@end table + +@section Advanced options + +@table @option +@item -map input stream id[:input stream id] +Set stream mapping from input streams to output streams. +Just enumerate the input streams in the order you want them in the output. +[input stream id] sets the (input) stream to sync against. +@item -map_meta_data outfile:infile +Set meta data information of outfile from infile. +@item -debug +Print specific debug info. +@item -benchmark +Add timings for benchmarking. +@item -dump +Dump each input packet. +@item -hex +When dumping packets, also dump the payload. +@item -bitexact +Only use bit exact algorithms (for codec testing). +@item -ps size +Set packet size in bits. +@item -re +Read input at native frame rate. Mainly used to simulate a grab device. +@item -loop_input +Loop over the input stream. Currently it works only for image +streams. This option is used for automatic FFserver testing. +@item -loop_output number_of_times +Repeatedly loop output for formats that support looping such as animated GIF +(0 will loop the output infinitely). +@item -threads count +Thread count. +@item -vsync parameter +Video sync method. Video will be stretched/squeezed to match the timestamps, +it is done by duplicating and dropping frames. With -map you can select from +which stream the timestamps should be taken. You can leave either video or +audio unchanged and sync the remaining stream(s) to the unchanged one. +@item -async samples_per_second +Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps, +the parameter is the maximum samples per second by which the audio is changed. +-async 1 is a special case where only the start of the audio stream is corrected +without any later correction. +@end table + +@node FFmpeg formula evaluator +@section FFmpeg formula evaluator + +When evaluating a rate control string, FFmpeg uses an internal formula +evaluator. + +The following binary operators are available: @code{+}, @code{-}, +@code{*}, @code{/}, @code{^}. + +The following unary operators are available: @code{+}, @code{-}, +@code{(...)}. + +The following functions are available: +@table @var +@item sinh(x) +@item cosh(x) +@item tanh(x) +@item sin(x) +@item cos(x) +@item tan(x) +@item exp(x) +@item log(x) +@item squish(x) +@item gauss(x) +@item abs(x) +@item max(x, y) +@item min(x, y) +@item gt(x, y) +@item lt(x, y) +@item eq(x, y) +@item bits2qp(bits) +@item qp2bits(qp) +@end table + +The following constants are available: +@table @var +@item PI +@item E +@item iTex +@item pTex +@item tex +@item mv +@item fCode +@item iCount +@item mcVar +@item var +@item isI +@item isP +@item isB +@item avgQP +@item qComp +@item avgIITex +@item avgPITex +@item avgPPTex +@item avgBPTex +@item avgTex +@end table + +@c man end + +@ignore + +@setfilename ffmpeg +@settitle FFmpeg video converter + +@c man begin SEEALSO +ffserver(1), ffplay(1) and the HTML documentation of @file{ffmpeg}. +@c man end + +@c man begin AUTHOR +Fabrice Bellard +@c man end + +@end ignore + +@section Protocols + +The filename can be @file{-} to read from standard input or to write +to standard output. + +FFmpeg also handles many protocols specified with an URL syntax. + +Use 'ffmpeg -formats' to see a list of the supported protocols. + +The protocol @code{http:} is currently used only to communicate with +FFserver (see the FFserver documentation). When FFmpeg will be a +video player it will also be used for streaming :-) + +@chapter Tips + +@itemize +@item For streaming at very low bitrate application, use a low frame rate +and a small GOP size. This is especially true for RealVideo where +the Linux player does not seem to be very fast, so it can miss +frames. An example is: + +@example +ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm +@end example + +@item The parameter 'q' which is displayed while encoding is the current +quantizer. The value 1 indicates that a very good quality could +be achieved. The value 31 indicates the worst quality. If q=31 appears +too often, it means that the encoder cannot compress enough to meet +your bitrate. You must either increase the bitrate, decrease the +frame rate or decrease the frame size. + +@item If your computer is not fast enough, you can speed up the +compression at the expense of the compression ratio. You can use +'-me zero' to speed up motion estimation, and '-intra' to disable +motion estimation completely (you have only I-frames, which means it +is about as good as JPEG compression). + +@item To have very low audio bitrates, reduce the sampling frequency +(down to 22050 kHz for MPEG audio, 22050 or 11025 for AC3). + +@item To have a constant quality (but a variable bitrate), use the option +'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst +quality). + +@item When converting video files, you can use the '-sameq' option which +uses the same quality factor in the encoder as in the decoder. +It allows almost lossless encoding. + +@end itemize + + +@chapter external libraries + +FFmpeg can be hooked up with a number of external libraries to add support +for more formats. + +@section AMR + +AMR comes in two different flavors, WB and NB. FFmpeg can make use of the +AMR WB (floating-point mode) and the AMR NB (both floating-point and +fixed-point mode) reference decoders and encoders. + +@itemize + +@item For AMR WB floating-point download TS26.204 V5.1.0 from +@url{http://www.3gpp.org/ftp/Specs/archive/26_series/26.204/26204-510.zip} +and extract the source to @file{libavcodec/amrwb_float/}. + +@item For AMR NB floating-point download TS26.104 REL-5 V5.1.0 from +@url{http://www.3gpp.org/ftp/Specs/archive/26_series/26.104/26104-510.zip} +and extract the source to @file{libavcodec/amr_float/}. +If you try this on Alpha, you may need to change @code{Word32} to +@code{int} in @file{amr/typedef.h}. + +@item For AMR NB fixed-point download TS26.073 REL-5 V5.1.0 from +@url{http://www.3gpp.org/ftp/Specs/archive/26_series/26.073/26073-510.zip} +and extract the source to @file{libavcodec/amr}. +You must also add @code{-DMMS_IO} and remove @code{-pedantic-errors} +to/from @code{CFLAGS} in @file{libavcodec/amr/makefile}, i.e. +``@code{CFLAGS = -Wall -I. \$(CFLAGS_\$(MODE)) -D\$(VAD) -DMMS_IO}''. + +@end itemize + + +@chapter Supported File Formats and Codecs + +You can use the @code{-formats} option to have an exhaustive list. + +@section File Formats + +FFmpeg supports the following file formats through the @code{libavformat} +library: + +@multitable @columnfractions .4 .1 .1 .4 +@item Supported File Format @tab Encoding @tab Decoding @tab Comments +@item MPEG audio @tab X @tab X +@item MPEG-1 systems @tab X @tab X +@tab muxed audio and video +@item MPEG-2 PS @tab X @tab X +@tab also known as @code{VOB} file +@item MPEG-2 TS @tab @tab X +@tab also known as DVB Transport Stream +@item ASF@tab X @tab X +@item AVI@tab X @tab X +@item WAV@tab X @tab X +@item Macromedia Flash@tab X @tab X +@tab Only embedded audio is decoded. +@item FLV @tab X @tab X +@tab Macromedia Flash video files +@item Real Audio and Video @tab X @tab X +@item Raw AC3 @tab X @tab X +@item Raw MJPEG @tab X @tab X +@item Raw MPEG video @tab X @tab X +@item Raw PCM8/16 bits, mulaw/Alaw@tab X @tab X +@item Raw CRI ADX audio @tab X @tab X +@item Raw Shorten audio @tab @tab X +@item SUN AU format @tab X @tab X +@item NUT @tab X @tab X @tab NUT Open Container Format +@item QuickTime @tab X @tab X +@item MPEG-4 @tab X @tab X +@tab MPEG-4 is a variant of QuickTime. +@item Raw MPEG4 video @tab X @tab X +@item DV @tab X @tab X +@item 4xm @tab @tab X +@tab 4X Technologies format, used in some games. +@item Playstation STR @tab @tab X +@item Id RoQ @tab @tab X +@tab Used in Quake III, Jedi Knight 2, other computer games. +@item Interplay MVE @tab @tab X +@tab Format used in various Interplay computer games. +@item WC3 Movie @tab @tab X +@tab Multimedia format used in Origin's Wing Commander III computer game. +@item Sega FILM/CPK @tab @tab X +@tab Used in many Sega Saturn console games. +@item Westwood Studios VQA/AUD @tab @tab X +@tab Multimedia formats used in Westwood Studios games. +@item Id Cinematic (.cin) @tab @tab X +@tab Used in Quake II. +@item FLIC format @tab @tab X +@tab .fli/.flc files +@item Sierra VMD @tab @tab X +@tab Used in Sierra CD-ROM games. +@item Sierra Online @tab @tab X +@tab .sol files used in Sierra Online games. +@item Matroska @tab @tab X +@item Electronic Arts Multimedia @tab @tab X +@tab Used in various EA games; files have extensions like WVE and UV2. +@item Nullsoft Video (NSV) format @tab @tab X +@item ADTS AAC audio @tab X @tab X +@item Creative VOC @tab X @tab X @tab Created for the Sound Blaster Pro. +@item American Laser Games MM @tab @tab X +@tab Multimedia format used in games like Mad Dog McCree +@item AVS @tab @tab X +@tab Multimedia format used by the Creature Shock game. +@item Smacker @tab @tab X +@tab Multimedia format used by many games. +@item GXF @tab X @tab X +@tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley playout servers. +@item CIN @tab @tab X +@tab Multimedia format used by Delphine Software games. +@item MXF @tab @tab X +@tab Material eXchange Format SMPTE 377M, used by D-Cinema, broadcast industry. +@item SEQ @tab @tab X +@tab Tiertex .seq files used in the DOS CDROM version of the game Flashback. +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@section Image Formats + +FFmpeg can read and write images for each frame of a video sequence. The +following image formats are supported: + +@multitable @columnfractions .4 .1 .1 .4 +@item Supported Image Format @tab Encoding @tab Decoding @tab Comments +@item PGM, PPM @tab X @tab X +@item PAM @tab X @tab X @tab PAM is a PNM extension with alpha support. +@item PGMYUV @tab X @tab X @tab PGM with U and V components in YUV 4:2:0 +@item JPEG @tab X @tab X @tab Progressive JPEG is not supported. +@item .Y.U.V @tab X @tab X @tab one raw file per component +@item animated GIF @tab X @tab X @tab Only uncompressed GIFs are generated. +@item PNG @tab X @tab X @tab 2 bit and 4 bit/pixel not supported yet. +@item Targa @tab @tab X @tab Targa (.TGA) image format. +@item TIFF @tab @tab X @tab Only 24 bit/pixel images are supported. +@item SGI @tab X @tab X @tab SGI RGB image format +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@section Video Codecs + +@multitable @columnfractions .4 .1 .1 .4 +@item Supported Codec @tab Encoding @tab Decoding @tab Comments +@item MPEG-1 video @tab X @tab X +@item MPEG-2 video @tab X @tab X +@item MPEG-4 @tab X @tab X +@item MSMPEG4 V1 @tab X @tab X +@item MSMPEG4 V2 @tab X @tab X +@item MSMPEG4 V3 @tab X @tab X +@item WMV7 @tab X @tab X +@item WMV8 @tab X @tab X @tab not completely working +@item WMV9 @tab @tab X @tab not completely working +@item VC1 @tab @tab X +@item H.261 @tab X @tab X +@item H.263(+) @tab X @tab X @tab also known as RealVideo 1.0 +@item H.264 @tab @tab X +@item RealVideo 1.0 @tab X @tab X +@item RealVideo 2.0 @tab X @tab X +@item MJPEG @tab X @tab X +@item lossless MJPEG @tab X @tab X +@item JPEG-LS @tab X @tab X @tab fourcc: MJLS, lossless and near-lossless is supported +@item Apple MJPEG-B @tab @tab X +@item Sunplus MJPEG @tab @tab X @tab fourcc: SP5X +@item DV @tab X @tab X +@item HuffYUV @tab X @tab X +@item FFmpeg Video 1 @tab X @tab X @tab experimental lossless codec (fourcc: FFV1) +@item FFmpeg Snow @tab X @tab X @tab experimental wavelet codec (fourcc: SNOW) +@item Asus v1 @tab X @tab X @tab fourcc: ASV1 +@item Asus v2 @tab X @tab X @tab fourcc: ASV2 +@item Creative YUV @tab @tab X @tab fourcc: CYUV +@item Sorenson Video 1 @tab X @tab X @tab fourcc: SVQ1 +@item Sorenson Video 3 @tab @tab X @tab fourcc: SVQ3 +@item On2 VP3 @tab @tab X @tab still experimental +@item On2 VP5 @tab @tab X @tab fourcc: VP50 +@item On2 VP6 @tab @tab X @tab fourcc: VP62 +@item Theora @tab @tab X @tab still experimental +@item Intel Indeo 3 @tab @tab X +@item FLV @tab X @tab X @tab Sorenson H.263 used in Flash +@item Flash Screen Video @tab @tab X @tab fourcc: FSV1 +@item ATI VCR1 @tab @tab X @tab fourcc: VCR1 +@item ATI VCR2 @tab @tab X @tab fourcc: VCR2 +@item Cirrus Logic AccuPak @tab @tab X @tab fourcc: CLJR +@item 4X Video @tab @tab X @tab Used in certain computer games. +@item Sony Playstation MDEC @tab @tab X +@item Id RoQ @tab @tab X @tab Used in Quake III, Jedi Knight 2, other computer games. +@item Xan/WC3 @tab @tab X @tab Used in Wing Commander III .MVE files. +@item Interplay Video @tab @tab X @tab Used in Interplay .MVE files. +@item Apple Animation @tab @tab X @tab fourcc: 'rle ' +@item Apple Graphics @tab @tab X @tab fourcc: 'smc ' +@item Apple Video @tab @tab X @tab fourcc: rpza +@item Apple QuickDraw @tab @tab X @tab fourcc: qdrw +@item Cinepak @tab @tab X +@item Microsoft RLE @tab @tab X +@item Microsoft Video-1 @tab @tab X +@item Westwood VQA @tab @tab X +@item Id Cinematic Video @tab @tab X @tab Used in Quake II. +@item Planar RGB @tab @tab X @tab fourcc: 8BPS +@item FLIC video @tab @tab X +@item Duck TrueMotion v1 @tab @tab X @tab fourcc: DUCK +@item Duck TrueMotion v2 @tab @tab X @tab fourcc: TM20 +@item VMD Video @tab @tab X @tab Used in Sierra VMD files. +@item MSZH @tab @tab X @tab Part of LCL +@item ZLIB @tab X @tab X @tab Part of LCL, encoder experimental +@item TechSmith Camtasia @tab @tab X @tab fourcc: TSCC +@item IBM Ultimotion @tab @tab X @tab fourcc: ULTI +@item Miro VideoXL @tab @tab X @tab fourcc: VIXL +@item QPEG @tab @tab X @tab fourccs: QPEG, Q1.0, Q1.1 +@item LOCO @tab @tab X @tab +@item Winnov WNV1 @tab @tab X @tab +@item Autodesk Animator Studio Codec @tab @tab X @tab fourcc: AASC +@item Fraps FPS1 @tab @tab X @tab +@item CamStudio @tab @tab X @tab fourcc: CSCD +@item American Laser Games Video @tab @tab X @tab Used in games like Mad Dog McCree +@item ZMBV @tab @tab X @tab +@item AVS Video @tab @tab X @tab Video encoding used by the Creature Shock game. +@item Smacker Video @tab @tab X @tab Video encoding used in Smacker. +@item RTjpeg @tab @tab X @tab Video encoding used in NuppelVideo files. +@item KMVC @tab @tab X @tab Codec used in Worms games. +@item VMware Video @tab @tab X @tab Codec used in videos captured by VMware. +@item Cin Video @tab @tab X @tab Codec used in Delphine Software games. +@item Tiertex Seq Video @tab @tab X @tab Codec used in DOS CDROM FlashBack game. +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@section Audio Codecs + +@multitable @columnfractions .4 .1 .1 .1 .7 +@item Supported Codec @tab Encoding @tab Decoding @tab Comments +@item MPEG audio layer 2 @tab IX @tab IX +@item MPEG audio layer 1/3 @tab IX @tab IX +@tab MP3 encoding is supported through the external library LAME. +@item AC3 @tab IX @tab IX +@tab liba52 is used internally for decoding. +@item Vorbis @tab X @tab X +@item WMA V1/V2 @tab @tab X +@item AAC @tab X @tab X +@tab Supported through the external library libfaac/libfaad. +@item Microsoft ADPCM @tab X @tab X +@item MS IMA ADPCM @tab X @tab X +@item QT IMA ADPCM @tab @tab X +@item 4X IMA ADPCM @tab @tab X +@item G.726 ADPCM @tab X @tab X +@item Duck DK3 IMA ADPCM @tab @tab X +@tab Used in some Sega Saturn console games. +@item Duck DK4 IMA ADPCM @tab @tab X +@tab Used in some Sega Saturn console games. +@item Westwood Studios IMA ADPCM @tab @tab X +@tab Used in Westwood Studios games like Command and Conquer. +@item SMJPEG IMA ADPCM @tab @tab X +@tab Used in certain Loki game ports. +@item CD-ROM XA ADPCM @tab @tab X +@item CRI ADX ADPCM @tab X @tab X +@tab Used in Sega Dreamcast games. +@item Electronic Arts ADPCM @tab @tab X +@tab Used in various EA titles. +@item Creative ADPCM @tab @tab X +@tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2 +@item RA144 @tab @tab X +@tab Real 14400 bit/s codec +@item RA288 @tab @tab X +@tab Real 28800 bit/s codec +@item RADnet @tab X @tab IX +@tab Real low bitrate AC3 codec, liba52 is used for decoding. +@item AMR-NB @tab X @tab X +@tab Supported through an external library. +@item AMR-WB @tab X @tab X +@tab Supported through an external library. +@item DV audio @tab @tab X +@item Id RoQ DPCM @tab @tab X +@tab Used in Quake III, Jedi Knight 2, other computer games. +@item Interplay MVE DPCM @tab @tab X +@tab Used in various Interplay computer games. +@item Xan DPCM @tab @tab X +@tab Used in Origin's Wing Commander IV AVI files. +@item Sierra Online DPCM @tab @tab X +@tab Used in Sierra Online game audio files. +@item Apple MACE 3 @tab @tab X +@item Apple MACE 6 @tab @tab X +@item FLAC lossless audio @tab @tab X +@item Shorten lossless audio @tab @tab X +@item Apple lossless audio @tab @tab X +@tab QuickTime fourcc 'alac' +@item FFmpeg Sonic @tab X @tab X +@tab experimental lossy/lossless codec +@item Qdesign QDM2 @tab @tab X +@tab there are still some distortions +@item Real COOK @tab @tab X +@tab All versions except 5.1 are supported +@item DSP Group TrueSpeech @tab @tab X +@item True Audio (TTA) @tab @tab X +@item Smacker Audio @tab @tab X +@item WavPack Audio @tab @tab X +@item Cin Audio @tab @tab X +@tab Codec used in Delphine Software games. +@item Intel Music Coder @tab @tab X +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@code{I} means that an integer-only version is available, too (ensures high +performance on systems without hardware floating point support). + +@chapter Platform Specific information + +@section Linux + +FFmpeg should be compiled with at least GCC 2.95.3. GCC 3.2 is the +preferred compiler now for FFmpeg. All future optimizations will depend on +features only found in GCC 3.2. + +@section BSD + +BSD make will not build FFmpeg, you need to install and use GNU Make +(@file{gmake}). + +@section Windows + +@subsection Native Windows compilation + +@itemize +@item Install the current versions of MSYS and MinGW from +@url{http://www.mingw.org/}. You can find detailed installation +instructions in the download section and the FAQ. + +@item If you want to test the FFplay, also download +the MinGW development library of SDL 1.2.x +(@file{SDL-devel-1.2.x-mingw32.tar.gz}) from +@url{http://www.libsdl.org}. Unpack it in a temporary directory, and +unpack the archive @file{i386-mingw32msvc.tar.gz} in the MinGW tool +directory. Edit the @file{sdl-config} script so that it gives the +correct SDL directory when invoked. + +@item Extract the current version of FFmpeg. + +@item Start the MSYS shell (file @file{msys.bat}). + +@item Change to the FFmpeg directory and follow + the instructions of how to compile FFmpeg (file +@file{INSTALL}). Usually, launching @file{./configure} and @file{make} +suffices. If you have problems using SDL, verify that +@file{sdl-config} can be launched from the MSYS command line. + +@item You can install FFmpeg in @file{Program Files/FFmpeg} by typing +@file{make install}. Don't forget to copy @file{SDL.dll} to the place +you launch @file{ffplay} from. + +@end itemize + +Notes: +@itemize + +@item The target @file{make wininstaller} can be used to create a +Nullsoft based Windows installer for FFmpeg and FFplay. @file{SDL.dll} +must be copied to the FFmpeg directory in order to build the +installer. + +@item By using @code{./configure --enable-shared} when configuring FFmpeg, +you can build @file{avcodec.dll} and @file{avformat.dll}. With +@code{make install} you install the FFmpeg DLLs and the associated +headers in @file{Program Files/FFmpeg}. + +@item Visual C++ compatibility: If you used @code{./configure --enable-shared} +when configuring FFmpeg, FFmpeg tries to use the Microsoft Visual +C++ @code{lib} tool to build @code{avcodec.lib} and +@code{avformat.lib}. With these libraries you can link your Visual C++ +code directly with the FFmpeg DLLs (see below). + +@end itemize + +@subsection Visual C++ compatibility + +FFmpeg will not compile under Visual C++ -- and it has too many +dependencies on the GCC compiler to make a port viable. However, +if you want to use the FFmpeg libraries in your own applications, +you can still compile those applications using Visual C++. An +important restriction to this is that you have to use the +dynamically linked versions of the FFmpeg libraries (i.e. the +DLLs), and you have to make sure that Visual-C++-compatible +import libraries are created during the FFmpeg build process. + +This description of how to use the FFmpeg libraries with Visual C++ is +based on Visual C++ 2005 Express Edition Beta 2. If you have a different +version, you might have to modify the procedures slightly. + +Here are the step-by-step instructions for building the FFmpeg libraries +so they can be used with Visual C++: + +@enumerate + +@item Install Visual C++ (if you haven't done so already). + +@item Install MinGW and MSYS as described above. + +@item Add a call to @file{vcvars32.bat} (which sets up the environment +variables for the Visual C++ tools) as the first line of +@file{msys.bat}. The standard location for @file{vcvars32.bat} is +@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat}, +and the standard location for @file{msys.bat} is +@file{C:\msys\1.0\msys.bat}. If this corresponds to your setup, add the +following line as the first line of @file{msys.bat}: + +@code{call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"} + +@item Start the MSYS shell (file @file{msys.bat}) and type @code{link.exe}. +If you get a help message with the command line options of @code{link.exe}, +this means your environment variables are set up correctly, the +Microsoft linker is on the path and will be used by FFmpeg to +create Visual-C++-compatible import libraries. + +@item Extract the current version of FFmpeg and change to the FFmpeg directory. + +@item Type the command +@code{./configure --enable-shared --disable-static --enable-memalign-hack} +to configure and, if that didn't produce any errors, +type @code{make} to build FFmpeg. + +@item The subdirectories @file{libavformat}, @file{libavcodec}, and +@file{libavutil} should now contain the files @file{avformat.dll}, +@file{avformat.lib}, @file{avcodec.dll}, @file{avcodec.lib}, +@file{avutil.dll}, and @file{avutil.lib}, respectively. Copy the three +DLLs to your System32 directory (typically @file{C:\Windows\System32}). + +@end enumerate + +And here is how to use these libraries with Visual C++: + +@enumerate + +@item Create a new console application ("File / New / Project") and then +select "Win32 Console Application". On the appropriate page of the +Application Wizard, uncheck the "Precompiled headers" option. + +@item Write the source code for your application, or, for testing, just +copy the code from an existing sample application into the source file +that Visual C++ has already created for you. (Note that your source +filehas to have a @code{.cpp} extension; otherwise, Visual C++ won't +compile the FFmpeg headers correctly because in C mode, it doesn't +recognize the @code{inline} keyword.) For example, you can copy +@file{output_example.c} from the FFmpeg distribution (but you will +have to make minor modifications so the code will compile under +C++, see below). + +@item Open the "Project / Properties" dialog box. In the "Configuration" +combo box, select "All Configurations" so that the changes you make will +affect both debug and release builds. In the tree view on the left hand +side, select "C/C++ / General", then edit the "Additional Include +Directories" setting to contain the complete paths to the +@file{libavformat}, @file{libavcodec}, and @file{libavutil} +subdirectories of your FFmpeg directory. Note that the directories have +to be separated using semicolons. Now select "Linker / General" from the +tree view and edit the "Additional Library Directories" setting to +contain the same three directories. + +@item Still in the "Project / Properties" dialog box, select "Linker / Input" +from the tree view, then add the files @file{avformat.lib}, +@file{avcodec.lib}, and @file{avutil.lib} to the end of the "Additional +Dependencies". Note that the names of the libraries have to be separated +using spaces. + +@item Now, select "C/C++ / Code Generation" from the tree view. Select +"Debug" in the "Configuration" combo box. Make sure that "Runtime +Library" is set to "Multi-threaded Debug DLL". Then, select "Release" in +the "Configuration" combo box and make sure that "Runtime Library" is +set to "Multi-threaded DLL". + +@item Click "OK" to close the "Project / Properties" dialog box and build +the application. Hopefully, it should compile and run cleanly. If you +used @file{output_example.c} as your sample application, you will get a +few compiler errors, but they are easy to fix. The first type of error +occurs because Visual C++ doesn't allow an @code{int} to be converted to +an @code{enum} without a cast. To solve the problem, insert the required +casts (this error occurs once for a @code{CodecID} and once for a +@code{CodecType}). The second type of error occurs because C++ requires +the return value of @code{malloc} to be cast to the exact type of the +pointer it is being assigned to. Visual C++ will complain that, for +example, @code{(void *)} is being assigned to @code{(uint8_t *)} without +an explicit cast. So insert an explicit cast in these places to silence +the compiler. The third type of error occurs because the @code{snprintf} +library function is called @code{_snprintf} under Visual C++. So just +add an underscore to fix the problem. With these changes, +@file{output_example.c} should compile under Visual C++, and the +resulting executable should produce valid video files. + +@end enumerate + +@subsection Cross compilation for Windows with Linux + +You must use the MinGW cross compilation tools available at +@url{http://www.mingw.org/}. + +Then configure FFmpeg with the following options: +@example +./configure --enable-mingw32 --cross-prefix=i386-mingw32msvc- +@end example +(you can change the cross-prefix according to the prefix chosen for the +MinGW tools). + +Then you can easily test FFmpeg with Wine +(@url{http://www.winehq.com/}). + +@subsection Compilation under Cygwin + +Cygwin works very much like Unix. + +Just install your Cygwin with all the "Base" packages, plus the +following "Devel" ones: +@example +binutils, gcc-core, make, subversion +@end example + +Do not install binutils-20060709-1 (they are buggy on shared builds); +use binutils-20050610-1 instead. + +Then run + +@example +./configure --enable-static --disable-shared +@end example + +to make a static build or + +@example +./configure --enable-shared --disable-static +@end example + +to build shared libraries. + +If you want to build FFmpeg with additional libraries, download Cygwin +"Devel" packages for Ogg and Vorbis from any Cygwin packages repository +and/or SDL, xvid, faac, faad2 packages from Cygwin Ports, +(@url{http://cygwinports.dotsrc.org/}). + +@subsection Crosscompilation for Windows under Cygwin + +With Cygwin you can create Windows binaries that don't need the cygwin1.dll. + +Just install your Cygwin as explained before, plus these additional +"Devel" packages: +@example +gcc-mingw-core, mingw-runtime, mingw-zlib +@end example + +and add some special flags to your configure invocation. + +For a static build run +@example +./configure --enable-mingw32 --enable-memalign-hack --enable-static --disable-shared --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin +@end example + +and for a build with shared libraries +@example +./configure --enable-mingw32 --enable-memalign-hack --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin +@end example + +@section BeOS + +The configure script should guess the configuration itself. +Networking support is currently not finished. +errno issues fixed by Andrew Bachmann. + +Old stuff: + +François Revol - revol at free dot fr - April 2002 + +The configure script should guess the configuration itself, +however I still didn't test building on the net_server version of BeOS. + +FFserver is broken (needs poll() implementation). + +There are still issues with errno codes, which are negative in BeOS, and +that FFmpeg negates when returning. This ends up turning errors into +valid results, then crashes. +(To be fixed) + +@chapter Developers Guide + +@section API +@itemize @bullet +@item libavcodec is the library containing the codecs (both encoding and +decoding). Look at @file{libavcodec/apiexample.c} to see how to use it. + +@item libavformat is the library containing the file format handling (mux and +demux code for several formats). Look at @file{ffplay.c} to use it in a +player. See @file{output_example.c} to use it to generate audio or video +streams. + +@end itemize + +@section Integrating libavcodec or libavformat in your program + +You can integrate all the source code of the libraries to link them +statically to avoid any version problem. All you need is to provide a +'config.mak' and a 'config.h' in the parent directory. See the defines +generated by ./configure to understand what is needed. + +You can use libavcodec or libavformat in your commercial program, but +@emph{any patch you make must be published}. The best way to proceed is +to send your patches to the FFmpeg mailing list. + +@node Coding Rules +@section Coding Rules + +FFmpeg is programmed in the ISO C90 language with a few additional +features from ISO C99, namely: +@itemize @bullet +@item +the @samp{inline} keyword; +@item +@samp{//} comments; +@item +designated struct initializers (@samp{struct s x = @{ .i = 17 @};}) +@item +compound literals (@samp{x = (struct s) @{ 17, 23 @};}) +@end itemize + +These features are supported by all compilers we care about, so we won't +accept patches to remove their use unless they absolutely don't impair +clarity and performance. + +All code must compile with GCC 2.95 and GCC 3.3. Currently, FFmpeg also +compiles with several other compilers, such as the Compaq ccc compiler +or Sun Studio 9, and we would like to keep it that way unless it would +be exceedingly involved. To ensure compatibility, please don't use any +additional C99 features or GCC extensions. Especially watch out for: +@itemize @bullet +@item +mixing statements and declarations; +@item +@samp{long long} (use @samp{int64_t} instead); +@item +@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar; +@item +GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}). +@end itemize + +Indent size is 4. +The presentation is the one specified by 'indent -i4 -kr -nut'. +The TAB character is forbidden outside of Makefiles as is any +form of trailing whitespace. Commits containing either will be +rejected by the Subversion repository. + +Main priority in FFmpeg is simplicity and small code size (=less +bugs). + +Comments: Use the JavaDoc/Doxygen +format (see examples below) so that code documentation +can be generated automatically. All nontrivial functions should have a comment +above them explaining what the function does, even if it's just one sentence. +All structures and their member variables should be documented, too. +@example +/** + * @@file mpeg.c + * MPEG codec. + * @@author ... + */ + +/** + * Summary sentence. + * more text ... + * ... + */ +typedef struct Foobar@{ + int var1; /**< var1 description */ + int var2; ///< var2 description + /** var3 description */ + int var3; +@} Foobar; + +/** + * Summary sentence. + * more text ... + * ... + * @@param my_parameter description of my_parameter + * @@return return value description + */ +int myfunc(int my_parameter) +... +@end example + +fprintf and printf are forbidden in libavformat and libavcodec, +please use av_log() instead. + +@section Development Policy + +@enumerate +@item + You must not commit code which breaks FFmpeg! (Meaning unfinished but + enabled code which breaks compilation or compiles but does not work or + breaks the regression tests) + You can commit unfinished stuff (for testing etc), but it must be disabled + (#ifdef etc) by default so it does not interfere with other developers' + work. +@item + You don't have to over-test things. If it works for you, and you think it + should work for others, then commit. If your code has problems + (portability, triggers compiler bugs, unusual environment etc) they will be + reported and eventually fixed. +@item + Do not commit unrelated changes together, split them into self-contained + pieces. +@item + Do not change behavior of the program (renaming options etc) without + first discussing it on the ffmpeg-devel mailing list. Do not remove + functionality from the code. Just improve! + + Note: Redundant code can be removed. +@item + Do not commit changes to the build system (Makefiles, configure script) + which change behavior, defaults etc, without asking first. The same + applies to compiler warning fixes, trivial looking fixes and to code + maintained by other developers. We usually have a reason for doing things + the way we do. Send your changes as patches to the ffmpeg-devel mailing + list, and if the code maintainers say OK, you may commit. This does not + apply to files you wrote and/or maintain. +@item + We refuse source indentation and other cosmetic changes if they are mixed + with functional changes, such commits will be rejected and removed. Every + developer has his own indentation style, you should not change it. Of course + if you (re)write something, you can use your own style, even though we would + prefer if the indentation throughout FFmpeg was consistent (Many projects + force a given indentation style - we don't.). If you really need to make + indentation changes (try to avoid this), separate them strictly from real + changes. + + NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code, + then either do NOT change the indentation of the inner part within (don't + move it to the right)! or do so in a separate commit +@item + Always fill out the commit log message. Describe in a few lines what you + changed and why. You can refer to mailing list postings if you fix a + particular bug. Comments such as "fixed!" or "Changed it." are unacceptable. +@item + If you apply a patch by someone else, include the name and email address in + the log message. Since the ffmpeg-cvslog mailing list is publicly + archived you should add some SPAM protection to the email address. Send an + answer to ffmpeg-devel (or wherever you got the patch from) saying that + you applied the patch. +@item + Do NOT commit to code actively maintained by others without permission. + Send a patch to ffmpeg-devel instead. If noone answers within a reasonable + timeframe (12h for build failures and security fixes, 3 days small changes, + 1 week for big patches) then commit your patch if you think it's OK. + Also note, the maintainer can simply ask for more time to review! +@item + Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits + are sent there and reviewed by all the other developers. Bugs and possible + improvements or general questions regarding commits are discussed there. We + expect you to react if problems with your code are uncovered. +@item + Update the documentation if you change behavior or add features. If you are + unsure how best to do this, send a patch to ffmpeg-devel, the documentation + maintainer(s) will review and commit your stuff. +@item + Never write to unallocated memory, never write over the end of arrays, + always check values read from some untrusted source before using them + as array index or other risky things. +@item + Remember to check if you need to bump versions for the specific libav + parts (libavutil, libavcodec, libavformat) you are changing. You need + to change the version integer and the version string. + Incrementing the first component means no backward compatibility to + previous versions (e.g. removal of a function from the public API). + Incrementing the second component means backward compatible change + (e.g. addition of a function to the public API). + Incrementing the third component means a noteworthy binary compatible + change (e.g. encoder bug fix that matters for the decoder). +@item + If you add a new codec, remember to update the changelog, add it to + the supported codecs table in the documentation and bump the second + component of the @file{libavcodec} version number appropriately. If + it has a fourcc, add it to @file{libavformat/avienc.c}, even if it + is only a decoder. +@end enumerate + +We think our rules are not too hard. If you have comments, contact us. + +Note, these rules are mostly borrowed from the MPlayer project. + +@section Submitting patches + +First, (@pxref{Coding Rules}) above if you didn't yet. + +When you submit your patch, try to send a unified diff (diff '-up' +option). I cannot read other diffs :-) + +Also please do not submit patches which contain several unrelated changes. +Split them into individual self-contained patches; this makes reviewing +them much easier. + +Run the regression tests before submitting a patch so that you can +verify that there are no big problems. + +Patches should be posted as base64 encoded attachments (or any other +encoding which ensures that the patch won't be trashed during +transmission) to the ffmpeg-devel mailing list, see +@url{http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel} + +It also helps quite a bit if you tell us what the patch does (for example +'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant +and has no lrint()') + +We reply to all submitted patches and either apply or reject with some +explanation why, but sometimes we are quite busy so it can take a week or two. + +@section Regression tests + +Before submitting a patch (or committing to the repository), you should at least +test that you did not break anything. + +The regression tests build a synthetic video stream and a synthetic +audio stream. These are then encoded and decoded with all codecs or +formats. The CRC (or MD5) of each generated file is recorded in a +result file. A 'diff' is launched to compare the reference results and +the result file. + +The regression tests then go on to test the FFserver code with a +limited set of streams. It is important that this step runs correctly +as well. + +Run 'make test' to test all the codecs and formats. + +Run 'make fulltest' to test all the codecs, formats and FFserver. + +[Of course, some patches may change the results of the regression tests. In +this case, the reference results of the regression tests shall be modified +accordingly]. + +@bye diff --git a/contrib/ffmpeg/doc/ffmpeg_powerpc_performance_evaluation_howto.txt b/contrib/ffmpeg/doc/ffmpeg_powerpc_performance_evaluation_howto.txt new file mode 100644 index 000000000..2eb4ee71a --- /dev/null +++ b/contrib/ffmpeg/doc/ffmpeg_powerpc_performance_evaluation_howto.txt @@ -0,0 +1,172 @@ +FFmpeg & evaluating performance on the PowerPC Architecture HOWTO + +(c) 2003-2004 Romain Dolbeau <romain@dolbeau.org> + + + +I - Introduction + +The PowerPC architecture and its SIMD extension AltiVec offer some +interesting tools to evaluate performance and improve the code. +This document tries to explain how to use those tools with FFmpeg. + +The architecture itself offers two ways to evaluate the performance of +a given piece of code: + +1) The Time Base Registers (TBL) +2) The Performance Monitor Counter Registers (PMC) + +The first ones are always available, always active, but they're not very +accurate: the registers increment by one every four *bus* cycles. On +my 667 Mhz tiBook (ppc7450), this means once every twenty *processor* +cycles. So we won't use that. + +The PMC are much more useful: not only can they report cycle-accurate +timing, but they can also be used to monitor many other parameters, +such as the number of AltiVec stalls for every kind of instruction, +or instruction cache misses. The downside is that not all processors +support the PMC (all G3, all G4 and the 970 do support them), and +they're inactive by default - you need to activate them with a +dedicated tool. Also, the number of available PMC depends on the +procesor: the various 604 have 2, the various 75x (aka. G3) have 4, +and the various 74xx (aka G4) have 6. + +*WARNING*: The PowerPC 970 is not very well documented, and its PMC +registers are 64 bits wide. To properly notify the code, you *must* +tune for the 970 (using --tune=970), or the code will assume 32 bit +registers. + + +II - Enabling FFmpeg PowerPC performance support + +This needs to be done by hand. First, you need to configure FFmpeg as +usual, but add the "--powerpc-perf-enable" option. For instance: + +##### +./configure --prefix=/usr/local/ffmpeg-svn --cc=gcc-3.3 --tune=7450 --powerpc-perf-enable +##### + +This will configure FFmpeg to install inside /usr/local/ffmpeg-svn, +compiling with gcc-3.3 (you should try to use this one or a newer +gcc), and tuning for the PowerPC 7450 (i.e. the newer G4; as a rule of +thumb, those at 550Mhz and more). It will also enable the PMC. + +You may also edit the file "config.h" to enable the following line: + +##### +// #define ALTIVEC_USE_REFERENCE_C_CODE 1 +##### + +If you enable this line, then the code will not make use of AltiVec, +but will use the reference C code instead. This is useful to compare +performance between two versions of the code. + +Also, the number of enabled PMC is defined in "libavcodec/ppc/dsputil_ppc.h": + +##### +#define POWERPC_NUM_PMC_ENABLED 4 +##### + +If you have a G4 CPU, you can enable all 6 PMC. DO NOT enable more +PMC than available on your CPU! + +Then, simply compile FFmpeg as usual (make && make install). + + + +III - Using FFmpeg PowerPC performance support + +This FFmeg can be used exactly as usual. But before exiting, FFmpeg +will dump a per-function report that looks like this: + +##### +PowerPC performance report + Values are from the PMC registers, and represent whatever the + registers are set to record. + Function "gmc1_altivec" (pmc1): + min: 231 + max: 1339867 + avg: 558.25 (255302) + Function "gmc1_altivec" (pmc2): + min: 93 + max: 2164 + avg: 267.31 (255302) + Function "gmc1_altivec" (pmc3): + min: 72 + max: 1987 + avg: 276.20 (255302) +(...) +##### + +In this example, PMC1 was set to record CPU cycles, PMC2 was set to +record AltiVec Permute Stall Cycles, and PMC3 was set to record AltiVec +Issue Stalls. + +The function "gmc1_altivec" was monitored 255302 times, and the +minimum execution time was 231 processor cycles. The max and average +aren't much use, as it's very likely the OS interrupted execution for +reasons of its own :-( + +With the exact same settings and source file, but using the reference C +code we get: + +##### +PowerPC performance report + Values are from the PMC registers, and represent whatever the + registers are set to record. + Function "gmc1_altivec" (pmc1): + min: 592 + max: 2532235 + avg: 962.88 (255302) + Function "gmc1_altivec" (pmc2): + min: 0 + max: 33 + avg: 0.00 (255302) + Function "gmc1_altivec" (pmc3): + min: 0 + max: 350 + avg: 0.03 (255302) +(...) +##### + +592 cycles, so the fastest AltiVec execution is about 2.5x faster than +the fastest C execution in this example. It's not perfect but it's not +bad (well I wrote this function so I can't say otherwise :-). + +Once you have that kind of report, you can try to improve things by +finding what goes wrong and fixing it; in the example above, one +should try to diminish the number of AltiVec stalls, as this *may* +improve performance. + + + +IV) Enabling the PMC in Mac OS X + +This is easy. Use "Monster" and "monster". Those tools come from +Apple's CHUD package, and can be found hidden in the developer web +site & FTP site. "MONster" is the graphical application, use it to +generate a config file specifying what each register should +monitor. Then use the command-line application "monster" to use that +config file, and enjoy the results. + +Note that "MONster" can be used for many other things, but it's +documented by Apple, it's not my subject. + +If you are using CHUD 4.4.2 or later, you'll notice that MONster is +no longer available. It's been superseeded by Shark, where +configuration of PMCs is available as a plugin. + + + +V) Enabling the PMC on Linux + +On linux you may use oprofile from http://oprofile.sf.net, depending on the +version and the cpu you may need to apply a patch[1] to access a set of the +possibile counters from the userspace application. You can always define them +using the kernel interface /dev/oprofile/* . + +[1] http://dev.gentoo.org/~lu_zero/development/oprofile-g4-20060423.patch + +-- +Romain Dolbeau <romain@dolbeau.org> +Luca Barbato <lu_zero@gentoo.org> diff --git a/contrib/ffmpeg/doc/ffplay-doc.texi b/contrib/ffmpeg/doc/ffplay-doc.texi new file mode 100644 index 000000000..db08eb38f --- /dev/null +++ b/contrib/ffmpeg/doc/ffplay-doc.texi @@ -0,0 +1,104 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFplay Documentation +@titlepage +@sp 7 +@center @titlefont{FFplay Documentation} +@sp 3 +@end titlepage + + +@chapter Introduction + +@c man begin DESCRIPTION +FFplay is a very simple and portable media player using the FFmpeg +libraries and the SDL library. It is mostly used as a testbed for the +various FFmpeg APIs. +@c man end + +@chapter Invocation + +@section Syntax +@example +@c man begin SYNOPSIS +ffplay [options] @file{input_file} +@c man end +@end example + +@c man begin OPTIONS +@section Main options + +@table @option +@item -h +show help +@item -x width +force displayed width +@item -y height +force displayed height +@item -an +disable audio +@item -vn +disable video +@item -nodisp +disable graphical display +@item -f fmt +force format +@end table + +@section Advanced options +@table @option +@item -stats +Show the stream duration, the codec parameters, the current position in +the stream and the audio/video synchronisation drift. +@item -rtp_tcp +Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful +if you are streaming with the RTSP protocol. +@item -sync type +Set the master clock to audio (@code{type=audio}), video +(@code{type=video}) or external (@code{type=ext}). Default is audio. The +master clock is used to control audio-video synchronization. Most media +players use audio as master clock, but in some cases (streaming or high +quality broadcast) it is necessary to change that. This option is mainly +used for debugging purposes. +@end table + +@section While playing + +@table @key +@item q, ESC +quit + +@item f +toggle full screen + +@item p, SPC +pause + +@item a +cycle audio channel + +@item v +cycle video channel + +@item w +show audio waves +@end table + +@c man end + +@ignore + +@setfilename ffplay +@settitle FFplay media player + +@c man begin SEEALSO +ffmpeg(1), ffserver(1) and the html documentation of @file{ffmpeg}. +@c man end + +@c man begin AUTHOR +Fabrice Bellard +@c man end + +@end ignore + +@bye diff --git a/contrib/ffmpeg/doc/ffserver-doc.texi b/contrib/ffmpeg/doc/ffserver-doc.texi new file mode 100644 index 000000000..ed67bb6c0 --- /dev/null +++ b/contrib/ffmpeg/doc/ffserver-doc.texi @@ -0,0 +1,224 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFserver Documentation +@titlepage +@sp 7 +@center @titlefont{FFserver Documentation} +@sp 3 +@end titlepage + + +@chapter Introduction + +@c man begin DESCRIPTION +FFserver is a streaming server for both audio and video. It supports +several live feeds, streaming from files and time shifting on live feeds +(you can seek to positions in the past on each live feed, provided you +specify a big enough feed storage in ffserver.conf). + +This documentation covers only the streaming aspects of ffserver / +ffmpeg. All questions about parameters for ffmpeg, codec questions, +etc. are not covered here. Read @file{ffmpeg-doc.html} for more +information. +@c man end + +@chapter QuickStart + +[Contributed by Philip Gladstone, philip-ffserver at gladstonefamily dot net] + +@section What can this do? + +When properly configured and running, you can capture video and audio in real +time from a suitable capture card, and stream it out over the Internet to +either Windows Media Player or RealAudio player (with some restrictions). + +It can also stream from files, though that is currently broken. Very often, a +web server can be used to serve up the files just as well. + +It can stream prerecorded video from .ffm files, though it is somewhat tricky +to make it work correctly. + +@section What do I need? + +I use Linux on a 900MHz Duron with a cheapo Bt848 based TV capture card. I'm +using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true, +I needed some special drivers for my motherboard-based sound card.] + +I understand that FreeBSD systems work just fine as well. + +@section How do I make it work? + +First, build the kit. It *really* helps to have installed LAME first. Then when +you run the ffserver ./configure, make sure that you have the --enable-mp3lame +flag turned on. + +LAME is important as it allows for streaming audio to Windows Media Player. +Don't ask why the other audio types do not work. + +As a simple test, just run the following two command lines (assuming that you +have a V4L video capture card): + +@example +./ffserver -f doc/ffserver.conf & +./ffmpeg http://localhost:8090/feed1.ffm +@end example + +At this point you should be able to go to your Windows machine and fire up +Windows Media Player (WMP). Go to Open URL and enter + +@example + http://<linuxbox>:8090/test.asf +@end example + +You should (after a short delay) see video and hear audio. + +WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to +transfer the entire file before starting to play. +The same is true of AVI files. + +@section What happens next? + +You should edit the ffserver.conf file to suit your needs (in terms of +frame rates etc). Then install ffserver and ffmpeg, write a script to start +them up, and off you go. + +@section Troubleshooting + +@subsection I don't hear any audio, but video is fine. + +Maybe you didn't install LAME, or got your ./configure statement wrong. Check +the ffmpeg output to see if a line referring to MP3 is present. If not, then +your configuration was incorrect. If it is, then maybe your wiring is not +set up correctly. Maybe the sound card is not getting data from the right +input source. Maybe you have a really awful audio interface (like I do) +that only captures in stereo and also requires that one channel be flipped. +If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before +starting ffmpeg. + +@subsection The audio and video loose sync after a while. + +Yes, they do. + +@subsection After a long while, the video update rate goes way down in WMP. + +Yes, it does. Who knows why? + +@subsection WMP 6.4 behaves differently to WMP 7. + +Yes, it does. Any thoughts on this would be gratefully received. These +differences extend to embedding WMP into a web page. [There are two +object IDs that you can use: The old one, which does not play well, and +the new one, which does (both tested on the same system). However, +I suspect that the new one is not available unless you have installed WMP 7]. + +@section What else can it do? + +You can replay video from .ffm files that was recorded earlier. +However, there are a number of caveats, including the fact that the +ffserver parameters must match the original parameters used to record the +file. If they do not, then ffserver deletes the file before recording into it. +(Now that I write this, it seems broken). + +You can fiddle with many of the codec choices and encoding parameters, and +there are a bunch more parameters that you cannot control. Post a message +to the mailing list if there are some 'must have' parameters. Look in +ffserver.conf for a list of the currently available controls. + +It will automatically generate the ASX or RAM files that are often used +in browsers. These files are actually redirections to the underlying ASF +or RM file. The reason for this is that the browser often fetches the +entire file before starting up the external viewer. The redirection files +are very small and can be transferred quickly. [The stream itself is +often 'infinite' and thus the browser tries to download it and never +finishes.] + +@section Tips + +* When you connect to a live stream, most players (WMP, RA, etc) want to +buffer a certain number of seconds of material so that they can display the +signal continuously. However, ffserver (by default) starts sending data +in realtime. This means that there is a pause of a few seconds while the +buffering is being done by the player. The good news is that this can be +cured by adding a '?buffer=5' to the end of the URL. This means that the +stream should start 5 seconds in the past -- and so the first 5 seconds +of the stream are sent as fast as the network will allow. It will then +slow down to real time. This noticeably improves the startup experience. + +You can also add a 'Preroll 15' statement into the ffserver.conf that will +add the 15 second prebuffering on all requests that do not otherwise +specify a time. In addition, ffserver will skip frames until a key_frame +is found. This further reduces the startup delay by not transferring data +that will be discarded. + +* You may want to adjust the MaxBandwidth in the ffserver.conf to limit +the amount of bandwidth consumed by live streams. + +@section Why does the ?buffer / Preroll stop working after a time? + +It turns out that (on my machine at least) the number of frames successfully +grabbed is marginally less than the number that ought to be grabbed. This +means that the timestamp in the encoded data stream gets behind realtime. +This means that if you say 'Preroll 10', then when the stream gets 10 +or more seconds behind, there is no Preroll left. + +Fixing this requires a change in the internals of how timestamps are +handled. + +@section Does the @code{?date=} stuff work. + +Yes (subject to the limitation outlined above). Also note that whenever you +start ffserver, it deletes the ffm file (if any parameters have changed), +thus wiping out what you had recorded before. + +The format of the @code{?date=xxxxxx} is fairly flexible. You should use one +of the following formats (the 'T' is literal): + +@example +* YYYY-MM-DDTHH:MM:SS (localtime) +* YYYY-MM-DDTHH:MM:SSZ (UTC) +@end example + +You can omit the YYYY-MM-DD, and then it refers to the current day. However +note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this +may be in the future and so is unlikely to be useful. + +You use this by adding the ?date= to the end of the URL for the stream. +For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}. + +@chapter Invocation +@section Syntax +@example +@c man begin SYNOPSIS +ffserver [options] +@c man end +@end example + +@section Options +@c man begin OPTIONS +@table @option +@item -L +Print the license. +@item -h +Print the help. +@item -f configfile +Use @file{configfile} instead of @file{/etc/ffserver.conf}. +@end table +@c man end + +@ignore + +@setfilename ffsserver +@settitle FFserver video server + +@c man begin SEEALSO +ffmpeg(1), ffplay(1), the @file{ffmpeg/doc/ffserver.conf} example and +the HTML documentation of @file{ffmpeg}. +@c man end + +@c man begin AUTHOR +Fabrice Bellard +@c man end + +@end ignore + +@bye diff --git a/contrib/ffmpeg/doc/ffserver.conf b/contrib/ffmpeg/doc/ffserver.conf new file mode 100644 index 000000000..a3b3ff412 --- /dev/null +++ b/contrib/ffmpeg/doc/ffserver.conf @@ -0,0 +1,349 @@ +# Port on which the server is listening. You must select a different +# port from your standard HTTP web server if it is running on the same +# computer. +Port 8090 + +# Address on which the server is bound. Only useful if you have +# several network interfaces. +BindAddress 0.0.0.0 + +# Number of simultaneous requests that can be handled. Since FFServer +# is very fast, it is more likely that you will want to leave this high +# and use MaxBandwidth, below. +MaxClients 1000 + +# This the maximum amount of kbit/sec that you are prepared to +# consume when streaming to clients. +MaxBandwidth 1000 + +# Access log file (uses standard Apache log file format) +# '-' is the standard output. +CustomLog - + +# Suppress that if you want to launch ffserver as a daemon. +NoDaemon + + +################################################################## +# Definition of the live feeds. Each live feed contains one video +# and/or audio sequence coming from an ffmpeg encoder or another +# ffserver. This sequence may be encoded simultaneously with several +# codecs at several resolutions. + +<Feed feed1.ffm> + +# You must use 'ffmpeg' to send a live feed to ffserver. In this +# example, you can type: +# +# ffmpeg http://localhost:8090/feed1.ffm + +# ffserver can also do time shifting. It means that it can stream any +# previously recorded live stream. The request should contain: +# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify +# a path where the feed is stored on disk. You also specify the +# maximum size of the feed, where zero means unlimited. Default: +# File=/tmp/feed_name.ffm FileMaxSize=5M +File /tmp/feed1.ffm +FileMaxSize 200K + +# You could specify +# ReadOnlyFile /saved/specialvideo.ffm +# This marks the file as readonly and it will not be deleted or updated. + +# Specify launch in order to start ffmpeg automatically. +# First ffmpeg must be defined with an appropriate path if needed, +# after that options can follow, but avoid adding the http:// field +#Launch ffmpeg + +# Only allow connections from localhost to the feed. +ACL allow 127.0.0.1 + +</Feed> + + +################################################################## +# Now you can define each stream which will be generated from the +# original audio and video stream. Each format has a filename (here +# 'test1.mpg'). FFServer will send this stream when answering a +# request containing this filename. + +<Stream test1.mpg> + +# coming from live feed 'feed1' +Feed feed1.ffm + +# Format of the stream : you can choose among: +# mpeg : MPEG-1 multiplexed video and audio +# mpegvideo : only MPEG-1 video +# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec) +# ogg : Ogg format (Vorbis audio codec) +# rm : RealNetworks-compatible stream. Multiplexed audio and video. +# ra : RealNetworks-compatible stream. Audio only. +# mpjpeg : Multipart JPEG (works with Netscape without any plugin) +# jpeg : Generate a single JPEG image. +# asf : ASF compatible streaming (Windows Media Player format). +# swf : Macromedia Flash compatible stream +# avi : AVI format (MPEG-4 video, MPEG audio sound) +# master : special ffmpeg stream used to duplicate a server +Format mpeg + +# Bitrate for the audio stream. Codecs usually support only a few +# different bitrates. +AudioBitRate 32 + +# Number of audio channels: 1 = mono, 2 = stereo +AudioChannels 1 + +# Sampling frequency for audio. When using low bitrates, you should +# lower this frequency to 22050 or 11025. The supported frequencies +# depend on the selected audio codec. +AudioSampleRate 44100 + +# Bitrate for the video stream +VideoBitRate 64 + +# Ratecontrol buffer size +VideoBufferSize 40 + +# Number of frames per second +VideoFrameRate 3 + +# Size of the video frame: WxH (default: 160x128) +# The following abbreviations are defined: sqcif, qcif, cif, 4cif +VideoSize 160x128 + +# Transmit only intra frames (useful for low bitrates, but kills frame rate). +#VideoIntraOnly + +# If non-intra only, an intra frame is transmitted every VideoGopSize +# frames. Video synchronization can only begin at an intra frame. +VideoGopSize 12 + +# More MPEG-4 parameters +# VideoHighQuality +# Video4MotionVector + +# Choose your codecs: +#AudioCodec mp2 +#VideoCodec mpeg1video + +# Suppress audio +#NoAudio + +# Suppress video +#NoVideo + +#VideoQMin 3 +#VideoQMax 31 + +# Set this to the number of seconds backwards in time to start. Note that +# most players will buffer 5-10 seconds of video, and also you need to allow +# for a keyframe to appear in the data stream. +#Preroll 15 + +# ACL: + +# You can allow ranges of addresses (or single addresses) +#ACL ALLOW <first address> <last address> + +# You can deny ranges of addresses (or single addresses) +#ACL DENY <first address> <last address> + +# You can repeat the ACL allow/deny as often as you like. It is on a per +# stream basis. The first match defines the action. If there are no matches, +# then the default is the inverse of the last ACL statement. +# +# Thus 'ACL allow localhost' only allows access from localhost. +# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and +# allow everybody else. + +</Stream> + + +################################################################## +# Example streams + + +# Multipart JPEG + +#<Stream test.mjpg> +#Feed feed1.ffm +#Format mpjpeg +#VideoFrameRate 2 +#VideoIntraOnly +#NoAudio +#Strict -1 +#</Stream> + + +# Single JPEG + +#<Stream test.jpg> +#Feed feed1.ffm +#Format jpeg +#VideoFrameRate 2 +#VideoIntraOnly +##VideoSize 352x240 +#NoAudio +#Strict -1 +#</Stream> + + +# Flash + +#<Stream test.swf> +#Feed feed1.ffm +#Format swf +#VideoFrameRate 2 +#VideoIntraOnly +#NoAudio +#</Stream> + + +# ASF compatible + +<Stream test.asf> +Feed feed1.ffm +Format asf +VideoFrameRate 15 +VideoSize 352x240 +VideoBitRate 256 +VideoBufferSize 40 +VideoGopSize 30 +AudioBitRate 64 +StartSendOnKey +</Stream> + + +# MP3 audio + +#<Stream test.mp3> +#Feed feed1.ffm +#Format mp2 +#AudioCodec mp3 +#AudioBitRate 64 +#AudioChannels 1 +#AudioSampleRate 44100 +#NoVideo +#</Stream> + + +# Ogg Vorbis audio + +#<Stream test.ogg> +#Feed feed1.ffm +#Title "Stream title" +#AudioBitRate 64 +#AudioChannels 2 +#AudioSampleRate 44100 +#NoVideo +#</Stream> + + +# Real with audio only at 32 kbits + +#<Stream test.ra> +#Feed feed1.ffm +#Format rm +#AudioBitRate 32 +#NoVideo +#NoAudio +#</Stream> + + +# Real with audio and video at 64 kbits + +#<Stream test.rm> +#Feed feed1.ffm +#Format rm +#AudioBitRate 32 +#VideoBitRate 128 +#VideoFrameRate 25 +#VideoGopSize 25 +#NoAudio +#</Stream> + + +################################################################## +# A stream coming from a file: you only need to set the input +# filename and optionally a new format. Supported conversions: +# AVI -> ASF + +#<Stream file.rm> +#File "/usr/local/httpd/htdocs/tlive.rm" +#NoAudio +#</Stream> + +#<Stream file.asf> +#File "/usr/local/httpd/htdocs/test.asf" +#NoAudio +#Author "Me" +#Copyright "Super MegaCorp" +#Title "Test stream from disk" +#Comment "Test comment" +#</Stream> + + +################################################################## +# RTSP examples +# +# You can access this stream with the RTSP URL: +# rtsp://localhost:5454/test1-rtsp.mpg +# +# A non-standard RTSP redirector is also created. Its URL is: +# http://localhost:8090/test1-rtsp.rtsp + +#<Stream test1-rtsp.mpg> +#Format rtp +#File "/usr/local/httpd/htdocs/test1.mpg" +#</Stream> + + +################################################################## +# SDP/multicast examples +# +# If you want to send your stream in multicast, you must set the +# multicast address with MulticastAddress. The port and the TTL can +# also be set. +# +# An SDP file is automatically generated by ffserver by adding the +# 'sdp' extension to the stream name (here +# http://localhost:8090/test1-sdp.sdp). You should usually give this +# file to your player to play the stream. +# +# The 'NoLoop' option can be used to avoid looping when the stream is +# terminated. + +#<Stream test1-sdp.mpg> +#Format rtp +#File "/usr/local/httpd/htdocs/test1.mpg" +#MulticastAddress 224.124.0.1 +#MulticastPort 5000 +#MulticastTTL 16 +#NoLoop +#</Stream> + + +################################################################## +# Special streams + +# Server status + +<Stream stat.html> +Format status + +# Only allow local people to get the status +ACL allow localhost +ACL allow 192.168.0.0 192.168.255.255 + +#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico +</Stream> + + +# Redirect index.html to the appropriate site + +<Redirect index.html> +URL http://www.ffmpeg.org/ +</Redirect> + + diff --git a/contrib/ffmpeg/doc/hooks.texi b/contrib/ffmpeg/doc/hooks.texi new file mode 100644 index 000000000..15013547c --- /dev/null +++ b/contrib/ffmpeg/doc/hooks.texi @@ -0,0 +1,113 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Video Hook Documentation +@titlepage +@sp 7 +@center @titlefont{Video Hook Documentation} +@sp 3 +@end titlepage + + +@chapter Introduction + + +The video hook functionality is designed (mostly) for live video. It allows +the video to be modified or examined between the decoder and the encoder. + +Any number of hook modules can be placed inline, and they are run in the +order that they were specified on the ffmpeg command line. + +Three modules are provided and are described below. They are all intended to +be used as a base for your own modules. + +Modules are loaded using the -vhook option to ffmpeg. The value of this parameter +is a space separated list of arguments. The first is the module name, and the rest +are passed as arguments to the Configure function of the module. + +@section null.c + +This does nothing. Actually it converts the input image to RGB24 and then converts +it back again. This is meant as a sample that you can use to test your setup. + +@section fish.c + +This implements a 'fish detector'. Essentially it converts the image into HSV +space and tests whether more than a certain percentage of the pixels fall into +a specific HSV cuboid. If so, then the image is saved into a file for processing +by other bits of code. + +Why use HSV? It turns out that HSV cuboids represent a more compact range of +colors than would an RGB cuboid. + +@section imlib2.c + +This module implements a text overlay for a video image. Currently it +supports a fixed overlay or reading the text from a file. The string +is passed through strftime so that it is easy to imprint the date and +time onto the image. + +You may also overlay an image (even semi-transparent) like TV stations do. +You may move either the text or the image around your video to create +scrolling credits, for example. + +Text fonts are being looked for in a FONTPATH environment variable. + +Options: +@multitable @columnfractions .2 .8 +@item @option{-c <color>} @tab The color of the text +@item @option{-F <fontname>} @tab The font face and size +@item @option{-t <text>} @tab The text +@item @option{-f <filename>} @tab The filename to read text from +@item @option{-x <expresion>} @tab X coordinate of text or image +@item @option{-y <expresion>} @tab Y coordinate of text or image +@item @option{-i <filename>} @tab The filename to read a image from +@end multitable + +Expresions are functions of these variables: +@multitable @columnfractions .2 .8 +@item @var{N} @tab frame number (starting at zero) +@item @var{H} @tab frame height +@item @var{W} @tab frame width +@item @var{h} @tab image height +@item @var{w} @tab image width +@item @var{X} @tab previous x coordinate of text or image +@item @var{Y} @tab previous y coordinate of text or image +@end multitable + +You may also use the constants @var{PI}, @var{E}, and the math functions available at the +FFmpeg formula evaluator at (@url{ffmpeg-doc.html#SEC13}), except @var{bits2qp(bits)} +and @var{qp2bits(qp)}. + +Usage examples: + +@example + # Remember to set the path to your fonts + FONTPATH="/cygdrive/c/WINDOWS/Fonts/" + FONTPATH="$FONTPATH:/usr/share/imlib2/data/fonts/" + FONTPATH="$FONTPATH:/usr/X11R6/lib/X11/fonts/TTF/" + export FONTPATH + + # Bulb dancing in a Lissajous pattern + ffmpeg -i input.avi -vhook \ + 'vhook/imlib2.dll -x W*(0.5+0.25*sin(N/47*PI))-w/2 -y H*(0.5+0.50*cos(N/97*PI))-h/2 -i /usr/share/imlib2/data/images/bulb.png' \ + -acodec copy -sameq output.avi + + # Text scrolling + ffmpeg -i input.avi -vhook \ + 'vhook/imlib2.dll -c red -F Vera.ttf/20 -x 150+0.5*N -y 70+0.25*N -t Hello' \ + -acodec copy -sameq output.avi +@end example + +@section ppm.c + +It's basically a launch point for a PPM pipe, so you can use any +executable (or script) which consumes a PPM on stdin and produces a PPM +on stdout (and flushes each frame). + +Usage example: + +@example +ffmpeg -i input -vhook "/path/to/ppm.so some-ppm-filter args" output +@end example + +@bye diff --git a/contrib/ffmpeg/doc/optimization.txt b/contrib/ffmpeg/doc/optimization.txt new file mode 100644 index 000000000..26c5ae64c --- /dev/null +++ b/contrib/ffmpeg/doc/optimization.txt @@ -0,0 +1,158 @@ +optimization Tips (for libavcodec): + +What to optimize: +If you plan to do non-x86 architecture specific optimizations (SIMD normally), +then take a look in the i386/ directory, as most important functions are +already optimized for MMX. + +If you want to do x86 optimizations then you can either try to finetune the +stuff in the i386 directory or find some other functions in the C source to +optimize, but there aren't many left. + +Understanding these overoptimized functions: +As many functions tend to be a bit difficult to understand because +of optimizations, it can be hard to optimize them further, or write +architecture-specific versions. It is recommened to look at older +revisions of the interesting files (for a web frontend try ViewVC at +http://svn.mplayerhq.hu/ffmpeg/trunk/). +Alternatively, look into the other architecture-specific versions in +the i386/, ppc/, alpha/ subdirectories. Even if you don't exactly +comprehend the instructions, it could help understanding the functions +and how they can be optimized. + +NOTE: If you still don't understand some function, ask at our mailing list!!! +(http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel) + + + +WTF is that function good for ....: +The primary purpose of that list is to avoid wasting time to optimize functions +which are rarely used + +put(_no_rnd)_pixels{,_x2,_y2,_xy2} + Used in motion compensation (en/decoding). + +avg_pixels{,_x2,_y2,_xy2} + Used in motion compensation of B-frames. + These are less important than the put*pixels functions. + +avg_no_rnd_pixels* + unused + +pix_abs16x16{,_x2,_y2,_xy2} + Used in motion estimation (encoding) with SAD. + +pix_abs8x8{,_x2,_y2,_xy2} + Used in motion estimation (encoding) with SAD of MPEG-4 4MV only. + These are less important than the pix_abs16x16* functions. + +put_mspel8_mc* / wmv2_mspel8* + Used only in WMV2. + it is not recommended that you waste your time with these, as WMV2 + is an ugly and relatively useless codec. + +mpeg4_qpel* / *qpel_mc* + Used in MPEG-4 qpel motion compensation (encoding & decoding). + The qpel8 functions are used only for 4mv, + the avg_* functions are used only for B-frames. + Optimizing them should have a significant impact on qpel + encoding & decoding. + +qpel{8,16}_mc??_old_c / *pixels{8,16}_l4 + Just used to work around a bug in an old libavcodec encoder version. + Don't optimize them. + +tpel_mc_func {put,avg}_tpel_pixels_tab + Used only for SVQ3, so only optimize them if you need fast SVQ3 decoding. + +add_bytes/diff_bytes + For huffyuv only, optimize if you want a faster ffhuffyuv codec. + +get_pixels / diff_pixels + Used for encoding, easy. + +clear_blocks + easiest to optimize + +gmc + Used for MPEG-4 gmc. + Optimizing this should have a significant effect on the gmc decoding + speed but it's very likely impossible to write in SIMD. + +gmc1 + Used for chroma blocks in MPEG-4 gmc with 1 warp point + (there are 4 luma & 2 chroma blocks per macroblock, so + only 1/3 of the gmc blocks use this, the other 2/3 + use the normal put_pixel* code, but only if there is + just 1 warp point). + Note: DivX5 gmc always uses just 1 warp point. + +pix_sum + Used for encoding. + +hadamard8_diff / sse / sad == pix_norm1 / dct_sad / quant_psnr / rd / bit + Specific compare functions used in encoding, it depends upon the + command line switches which of these are used. + Don't waste your time with dct_sad & quant_psnr, they aren't + really useful. + +put_pixels_clamped / add_pixels_clamped + Used for en/decoding in the IDCT, easy. + Note, some optimized IDCTs have the add/put clamped code included and + then put_pixels_clamped / add_pixels_clamped will be unused. + +idct/fdct + idct (encoding & decoding) + fdct (encoding) + difficult to optimize + +dct_quantize_trellis + Used for encoding with trellis quantization. + difficult to optimize + +dct_quantize + Used for encoding. + +dct_unquantize_mpeg1 + Used in MPEG-1 en/decoding. + +dct_unquantize_mpeg2 + Used in MPEG-2 en/decoding. + +dct_unquantize_h263 + Used in MPEG-4/H.263 en/decoding. + +FIXME remaining functions? +BTW, most of these functions are in dsputil.c/.h, some are in mpegvideo.c/.h. + + + +Alignment: +Some instructions on some architectures have strict alignment restrictions, +for example most SSE/SSE2 instructions on x86. +The minimum guaranteed alignment is written in the .h files, for example: + void (*put_pixels_clamped)(const DCTELEM *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size); + + + +Links: +http://www.aggregate.org/MAGIC/ + +x86-specific: +http://developer.intel.com/design/pentium4/manuals/248966.htm + +The IA-32 Intel Architecture Software Developer's Manual, Volume 2: +Instruction Set Reference +http://developer.intel.com/design/pentium4/manuals/245471.htm + +http://www.agner.org/assem/ + +AMD Athlon Processor x86 Code Optimization Guide: +http://www.amd.com/us-en/assets/content_type/white_papers_and_tech_docs/22007.pdf + +GCC asm links: +official doc but quite ugly +http://gcc.gnu.org/onlinedocs/gcc/Extended-Asm.html + +a bit old (note "+" is valid for input-output, even though the next disagrees) +http://www.cs.virginia.edu/~clc5q/gcc-inline-asm.pdf diff --git a/contrib/ffmpeg/doc/soc.txt b/contrib/ffmpeg/doc/soc.txt new file mode 100644 index 000000000..8b4a86db8 --- /dev/null +++ b/contrib/ffmpeg/doc/soc.txt @@ -0,0 +1,24 @@ +Google Summer of Code and similar project guidelines + +Summer of Code is a project by Google in which students are paid to implement +some nice new features for various participating open source projects ... + +This text is a collection of things to take care of for the next soc as +it's a little late for this year's soc (2006). + +The Goal: +Our goal in respect to soc is and must be of course exactly one thing and +that is to improve FFmpeg, to reach this goal, code must +* conform to the svn policy and patch submission guidelines +* must improve FFmpeg somehow (faster, smaller, "better", + more codecs supported, fewer bugs, cleaner, ...) + +for mentors and other developers to help students to reach that goal it is +essential that changes to their codebase are publicly visible, clean and +easy reviewable that again leads us to: +* use of a revision control system like svn +* separation of cosmetic from non-cosmetic changes (this is almost entirely + ignored by mentors and students in soc 2006 which might lead to a suprise + when the code will be reviewed at the end before a possible inclusion in + FFmpeg, individual changes were generally not reviewable due to cosmetics). +* frequent commits, so that comments can be provided early diff --git a/contrib/ffmpeg/doc/texi2pod.pl b/contrib/ffmpeg/doc/texi2pod.pl new file mode 100755 index 000000000..c414ffcc6 --- /dev/null +++ b/contrib/ffmpeg/doc/texi2pod.pl @@ -0,0 +1,427 @@ +#! /usr/bin/perl -w + +# Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc. + +# This file is part of GNU CC. + +# GNU CC is free software; you can redistribute it and/or modify +# it under the terms of the GNU General Public License as published by +# the Free Software Foundation; either version 2, or (at your option) +# any later version. + +# GNU CC is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# GNU General Public License for more details. + +# You should have received a copy of the GNU General Public License +# along with GNU CC; see the file COPYING. If not, write to +# the Free Software Foundation, 51 Franklin Street, Fifth Floor, +# Boston, MA 02110-1301 USA + +# This does trivial (and I mean _trivial_) conversion of Texinfo +# markup to Perl POD format. It's intended to be used to extract +# something suitable for a manpage from a Texinfo document. + +$output = 0; +$skipping = 0; +%sects = (); +$section = ""; +@icstack = (); +@endwstack = (); +@skstack = (); +@instack = (); +$shift = ""; +%defs = (); +$fnno = 1; +$inf = ""; +$ibase = ""; + +while ($_ = shift) { + if (/^-D(.*)$/) { + if ($1 ne "") { + $flag = $1; + } else { + $flag = shift; + } + $value = ""; + ($flag, $value) = ($flag =~ /^([^=]+)(?:=(.+))?/); + die "no flag specified for -D\n" + unless $flag ne ""; + die "flags may only contain letters, digits, hyphens, dashes and underscores\n" + unless $flag =~ /^[a-zA-Z0-9_-]+$/; + $defs{$flag} = $value; + } elsif (/^-/) { + usage(); + } else { + $in = $_, next unless defined $in; + $out = $_, next unless defined $out; + usage(); + } +} + +if (defined $in) { + $inf = gensym(); + open($inf, "<$in") or die "opening \"$in\": $!\n"; + $ibase = $1 if $in =~ m|^(.+)/[^/]+$|; +} else { + $inf = \*STDIN; +} + +if (defined $out) { + open(STDOUT, ">$out") or die "opening \"$out\": $!\n"; +} + +while(defined $inf) { +while(<$inf>) { + # Certain commands are discarded without further processing. + /^\@(?: + [a-z]+index # @*index: useful only in complete manual + |need # @need: useful only in printed manual + |(?:end\s+)?group # @group .. @end group: ditto + |page # @page: ditto + |node # @node: useful only in .info file + |(?:end\s+)?ifnottex # @ifnottex .. @end ifnottex: use contents + )\b/x and next; + + chomp; + + # Look for filename and title markers. + /^\@setfilename\s+([^.]+)/ and $fn = $1, next; + /^\@settitle\s+([^.]+)/ and $tl = postprocess($1), next; + + # Identify a man title but keep only the one we are interested in. + /^\@c\s+man\s+title\s+([A-Za-z0-9-]+)\s+(.+)/ and do { + if (exists $defs{$1}) { + $fn = $1; + $tl = postprocess($2); + } + next; + }; + + # Look for blocks surrounded by @c man begin SECTION ... @c man end. + # This really oughta be @ifman ... @end ifman and the like, but such + # would require rev'ing all other Texinfo translators. + /^\@c\s+man\s+begin\s+([A-Z]+)\s+([A-Za-z0-9-]+)/ and do { + $output = 1 if exists $defs{$2}; + $sect = $1; + next; + }; + /^\@c\s+man\s+begin\s+([A-Z]+)/ and $sect = $1, $output = 1, next; + /^\@c\s+man\s+end/ and do { + $sects{$sect} = "" unless exists $sects{$sect}; + $sects{$sect} .= postprocess($section); + $section = ""; + $output = 0; + next; + }; + + # handle variables + /^\@set\s+([a-zA-Z0-9_-]+)\s*(.*)$/ and do { + $defs{$1} = $2; + next; + }; + /^\@clear\s+([a-zA-Z0-9_-]+)/ and do { + delete $defs{$1}; + next; + }; + + next unless $output; + + # Discard comments. (Can't do it above, because then we'd never see + # @c man lines.) + /^\@c\b/ and next; + + # End-block handler goes up here because it needs to operate even + # if we are skipping. + /^\@end\s+([a-z]+)/ and do { + # Ignore @end foo, where foo is not an operation which may + # cause us to skip, if we are presently skipping. + my $ended = $1; + next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex)$/; + + die "\@end $ended without \@$ended at line $.\n" unless defined $endw; + die "\@$endw ended by \@end $ended at line $.\n" unless $ended eq $endw; + + $endw = pop @endwstack; + + if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex)$/) { + $skipping = pop @skstack; + next; + } elsif ($ended =~ /^(?:example|smallexample|display)$/) { + $shift = ""; + $_ = ""; # need a paragraph break + } elsif ($ended =~ /^(?:itemize|enumerate|[fv]?table)$/) { + $_ = "\n=back\n"; + $ic = pop @icstack; + } else { + die "unknown command \@end $ended at line $.\n"; + } + }; + + # We must handle commands which can cause skipping even while we + # are skipping, otherwise we will not process nested conditionals + # correctly. + /^\@ifset\s+([a-zA-Z0-9_-]+)/ and do { + push @endwstack, $endw; + push @skstack, $skipping; + $endw = "ifset"; + $skipping = 1 unless exists $defs{$1}; + next; + }; + + /^\@ifclear\s+([a-zA-Z0-9_-]+)/ and do { + push @endwstack, $endw; + push @skstack, $skipping; + $endw = "ifclear"; + $skipping = 1 if exists $defs{$1}; + next; + }; + + /^\@(ignore|menu|iftex)\b/ and do { + push @endwstack, $endw; + push @skstack, $skipping; + $endw = $1; + $skipping = 1; + next; + }; + + next if $skipping; + + # Character entities. First the ones that can be replaced by raw text + # or discarded outright: + s/\@copyright\{\}/(c)/g; + s/\@dots\{\}/.../g; + s/\@enddots\{\}/..../g; + s/\@([.!? ])/$1/g; + s/\@[:-]//g; + s/\@bullet(?:\{\})?/*/g; + s/\@TeX\{\}/TeX/g; + s/\@pounds\{\}/\#/g; + s/\@minus(?:\{\})?/-/g; + s/\\,/,/g; + + # Now the ones that have to be replaced by special escapes + # (which will be turned back into text by unmunge()) + s/&/&/g; + s/\@\{/{/g; + s/\@\}/}/g; + s/\@\@/&at;/g; + + # Inside a verbatim block, handle @var specially. + if ($shift ne "") { + s/\@var\{([^\}]*)\}/<$1>/g; + } + + # POD doesn't interpret E<> inside a verbatim block. + if ($shift eq "") { + s/</</g; + s/>/>/g; + } else { + s/</</g; + s/>/>/g; + } + + # Single line command handlers. + + /^\@include\s+(.+)$/ and do { + push @instack, $inf; + $inf = gensym(); + + # Try cwd and $ibase. + open($inf, "<" . $1) + or open($inf, "<" . $ibase . "/" . $1) + or die "cannot open $1 or $ibase/$1: $!\n"; + next; + }; + + /^\@(?:section|unnumbered|unnumberedsec|center)\s+(.+)$/ + and $_ = "\n=head2 $1\n"; + /^\@subsection\s+(.+)$/ + and $_ = "\n=head3 $1\n"; + + # Block command handlers: + /^\@itemize\s+(\@[a-z]+|\*|-)/ and do { + push @endwstack, $endw; + push @icstack, $ic; + $ic = $1; + $_ = "\n=over 4\n"; + $endw = "itemize"; + }; + + /^\@enumerate(?:\s+([a-zA-Z0-9]+))?/ and do { + push @endwstack, $endw; + push @icstack, $ic; + if (defined $1) { + $ic = $1 . "."; + } else { + $ic = "1."; + } + $_ = "\n=over 4\n"; + $endw = "enumerate"; + }; + + /^\@([fv]?table)\s+(\@[a-z]+)/ and do { + push @endwstack, $endw; + push @icstack, $ic; + $endw = $1; + $ic = $2; + $ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env)/B/; + $ic =~ s/\@(?:code|kbd)/C/; + $ic =~ s/\@(?:dfn|var|emph|cite|i)/I/; + $ic =~ s/\@(?:file)/F/; + $_ = "\n=over 4\n"; + }; + + /^\@((?:small)?example|display)/ and do { + push @endwstack, $endw; + $endw = $1; + $shift = "\t"; + $_ = ""; # need a paragraph break + }; + + /^\@itemx?\s*(.+)?$/ and do { + if (defined $1) { + # Entity escapes prevent munging by the <> processing below. + $_ = "\n=item $ic\<$1\>\n"; + } else { + $_ = "\n=item $ic\n"; + $ic =~ y/A-Ya-y/B-Zb-z/; + $ic =~ s/(\d+)/$1 + 1/eg; + } + }; + + $section .= $shift.$_."\n"; +} +# End of current file. +close($inf); +$inf = pop @instack; +} + +die "No filename or title\n" unless defined $fn && defined $tl; + +$sects{NAME} = "$fn \- $tl\n"; +$sects{FOOTNOTES} .= "=back\n" if exists $sects{FOOTNOTES}; + +for $sect (qw(NAME SYNOPSIS DESCRIPTION OPTIONS EXAMPLES ENVIRONMENT FILES + BUGS NOTES FOOTNOTES SEEALSO AUTHOR COPYRIGHT)) { + if(exists $sects{$sect}) { + $head = $sect; + $head =~ s/SEEALSO/SEE ALSO/; + print "=head1 $head\n\n"; + print scalar unmunge ($sects{$sect}); + print "\n"; + } +} + +sub usage +{ + die "usage: $0 [-D toggle...] [infile [outfile]]\n"; +} + +sub postprocess +{ + local $_ = $_[0]; + + # @value{foo} is replaced by whatever 'foo' is defined as. + while (m/(\@value\{([a-zA-Z0-9_-]+)\})/g) { + if (! exists $defs{$2}) { + print STDERR "Option $2 not defined\n"; + s/\Q$1\E//; + } else { + $value = $defs{$2}; + s/\Q$1\E/$value/; + } + } + + # Formatting commands. + # Temporary escape for @r. + s/\@r\{([^\}]*)\}/R<$1>/g; + s/\@(?:dfn|var|emph|cite|i)\{([^\}]*)\}/I<$1>/g; + s/\@(?:code|kbd)\{([^\}]*)\}/C<$1>/g; + s/\@(?:gccoptlist|samp|strong|key|option|env|command|b)\{([^\}]*)\}/B<$1>/g; + s/\@sc\{([^\}]*)\}/\U$1/g; + s/\@file\{([^\}]*)\}/F<$1>/g; + s/\@w\{([^\}]*)\}/S<$1>/g; + s/\@(?:dmn|math)\{([^\}]*)\}/$1/g; + + # Cross references are thrown away, as are @noindent and @refill. + # (@noindent is impossible in .pod, and @refill is unnecessary.) + # @* is also impossible in .pod; we discard it and any newline that + # follows it. Similarly, our macro @gol must be discarded. + + s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g; + s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g; + s/;\s+\@pxref\{(?:[^\}]*)\}//g; + s/\@noindent\s*//g; + s/\@refill//g; + s/\@gol//g; + s/\@\*\s*\n?//g; + + # @uref can take one, two, or three arguments, with different + # semantics each time. @url and @email are just like @uref with + # one argument, for our purposes. + s/\@(?:uref|url|email)\{([^\},]*)\}/<B<$1>>/g; + s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g; + s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g; + + # Turn B<blah I<blah> blah> into B<blah> I<blah> B<blah> to + # match Texinfo semantics of @emph inside @samp. Also handle @r + # inside bold. + s/</</g; + s/>/>/g; + 1 while s/B<((?:[^<>]|I<[^<>]*>)*)R<([^>]*)>/B<$1>${2}B</g; + 1 while (s/B<([^<>]*)I<([^>]+)>/B<$1>I<$2>B</g); + 1 while (s/I<([^<>]*)B<([^>]+)>/I<$1>B<$2>I</g); + s/[BI]<>//g; + s/([BI])<(\s+)([^>]+)>/$2$1<$3>/g; + s/([BI])<([^>]+?)(\s+)>/$1<$2>$3/g; + + # Extract footnotes. This has to be done after all other + # processing because otherwise the regexp will choke on formatting + # inside @footnote. + while (/\@footnote/g) { + s/\@footnote\{([^\}]+)\}/[$fnno]/; + add_footnote($1, $fnno); + $fnno++; + } + + return $_; +} + +sub unmunge +{ + # Replace escaped symbols with their equivalents. + local $_ = $_[0]; + + s/</E<lt>/g; + s/>/E<gt>/g; + s/{/\{/g; + s/}/\}/g; + s/&at;/\@/g; + s/&/&/g; + return $_; +} + +sub add_footnote +{ + unless (exists $sects{FOOTNOTES}) { + $sects{FOOTNOTES} = "\n=over 4\n\n"; + } + + $sects{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++; + $sects{FOOTNOTES} .= $_[0]; + $sects{FOOTNOTES} .= "\n\n"; +} + +# stolen from Symbol.pm +{ + my $genseq = 0; + sub gensym + { + my $name = "GEN" . $genseq++; + my $ref = \*{$name}; + delete $::{$name}; + return $ref; + } +} |