diff options
author | theuni <theuni-nospam-@xbmc.org> | 2011-01-24 16:05:21 -0500 |
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committer | theuni <theuni-nospam-@xbmc.org> | 2011-01-24 16:05:21 -0500 |
commit | c51b1189e3d5353e842991f5859ddcea0f73e426 (patch) | |
tree | ef2cb8a6184699aa614f3655dca4ce661cdc108e /lib/ffmpeg/doc | |
parent | be61ebdc9e897fe40c6f371111724de79ddee8d5 (diff) |
Merged cptspiff's code-reshuffle branch.
Squashed commit due to build breakage during code-reshuffle history.
Conflicts:
xbmc/Util.cpp
xbmc/cdrip/CDDARipper.cpp
xbmc/filesystem/Directory.cpp
xbmc/filesystem/File.cpp
Diffstat (limited to 'lib/ffmpeg/doc')
-rw-r--r-- | lib/ffmpeg/doc/APIchanges | 297 | ||||
-rw-r--r-- | lib/ffmpeg/doc/TODO | 82 | ||||
-rw-r--r-- | lib/ffmpeg/doc/avutil.txt | 37 | ||||
-rw-r--r-- | lib/ffmpeg/doc/developer.texi | 436 | ||||
-rw-r--r-- | lib/ffmpeg/doc/faq.texi | 501 | ||||
-rw-r--r-- | lib/ffmpeg/doc/ffmpeg-doc.texi | 984 | ||||
-rw-r--r-- | lib/ffmpeg/doc/ffplay-doc.texi | 173 | ||||
-rw-r--r-- | lib/ffmpeg/doc/ffprobe-doc.texi | 123 | ||||
-rw-r--r-- | lib/ffmpeg/doc/ffserver-doc.texi | 276 | ||||
-rw-r--r-- | lib/ffmpeg/doc/ffserver.conf | 377 | ||||
-rw-r--r-- | lib/ffmpeg/doc/fftools-common-opts.texi | 89 | ||||
-rw-r--r-- | lib/ffmpeg/doc/filters.texi | 258 | ||||
-rw-r--r-- | lib/ffmpeg/doc/general.texi | 1067 | ||||
-rw-r--r-- | lib/ffmpeg/doc/issue_tracker.txt | 228 | ||||
-rw-r--r-- | lib/ffmpeg/doc/libavfilter.texi | 104 | ||||
-rw-r--r-- | lib/ffmpeg/doc/optimization.txt | 235 | ||||
-rw-r--r-- | lib/ffmpeg/doc/rate_distortion.txt | 61 | ||||
-rw-r--r-- | lib/ffmpeg/doc/snow.txt | 630 | ||||
-rw-r--r-- | lib/ffmpeg/doc/soc.txt | 24 | ||||
-rw-r--r-- | lib/ffmpeg/doc/swscale.txt | 99 | ||||
-rw-r--r-- | lib/ffmpeg/doc/tablegen.txt | 70 | ||||
-rwxr-xr-x | lib/ffmpeg/doc/texi2pod.pl | 423 | ||||
-rw-r--r-- | lib/ffmpeg/doc/viterbi.txt | 110 |
23 files changed, 6684 insertions, 0 deletions
diff --git a/lib/ffmpeg/doc/APIchanges b/lib/ffmpeg/doc/APIchanges new file mode 100644 index 0000000000..2150225343 --- /dev/null +++ b/lib/ffmpeg/doc/APIchanges @@ -0,0 +1,297 @@ +Never assume the API of libav* to be stable unless at least 1 week has passed since +the last major version increase. +The last version increases were: +libavcodec: ? +libavdevice: ? +libavfilter: 2009-10-18 +libavformat: ? +libpostproc: ? +libswscale: ? +libavutil: 2009-03-08 + + +API changes, most recent first: + +2010-07-11 - r24199 - lavc 52.83.0 + Add AVCodecContext.lpc_type and AVCodecContext.lpc_passes fields. + Add AVLPCType enum. + Deprecate AVCodecContext.use_lpc. + +2010-07-11 - r24185 - lavc 52.82.0 - avsubtitle_free() + Add a function for free the contents of a AVSubtitle generated by + avcodec_decode_subtitle. + +2010-07-11 - r24174 - lavu 50.22.0 - bswap.h and intreadwrite.h + Make the bswap.h and intreadwrite.h API public. + +2010-07-08 - r24101 - lavu 50.21.0 - pixdesc.h + Rename read/write_line() to av_read/write_image_line(). + +2010-07-07 - r24091 - lavfi 1.21.0 - avfilter_copy_picref_props() + Add avfilter_copy_picref_props(). + +2010-07-03 - r24021 - lavc 52.79.0 + Add FF_COMPLIANCE_UNOFFICIAL and change all instances of + FF_COMPLIANCE_INOFFICIAL to use FF_COMPLIANCE_UNOFFICIAL. + +2010-07-02 - r23985 - lavu 50.20.0 - lfg.h + Export av_lfg_init(), av_lfg_get(), av_mlfg_get(), and av_bmg_get() through + lfg.h. + +2010-06-28 - r23835 - lavfi 1.20.1 - av_parse_color() + Extend av_parse_color() syntax, make it accept an alpha value specifier and + set the alpha value to 255 by default. + +2010-06-22 - r23706 - lavf 52.71.0 - URLProtocol.priv_data_size, priv_data_class + Add priv_data_size and priv_data_class to URLProtocol. + +2010-06-22 - r23704 - lavf 52.70.0 - url_alloc(), url_connect() + Add url_alloc() and url_connect(). + +2010-06-22 - r23702 - lavf 52.69.0 - av_register_protocol2() + Add av_register_protocol2(), deprecating av_register_protocol(). + +2010-06-09 - r23551 - lavu 50.19.0 - av_compare_mod() + Add av_compare_mod() to libavutil/mathematics.h. + +2010-06-05 - r23485 - lavu 50.18.0 - eval API + Make the eval API public. + +2010-06-04 - r23461 - lavu 50.17.0 - AV_BASE64_SIZE + Add AV_BASE64_SIZE() macro. + +2010-06-02 - r23421 - lavc 52.73.0 - av_get_codec_tag_string() + Add av_get_codec_tag_string(). + +2010-06-01 - r31301 - lsws 0.11.0 - convertPalette API + Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24(). + +2010-05-26 - r23334 - lavc 52.72.0 - CODEC_CAP_EXPERIMENTAL + Add CODEC_CAP_EXPERIMENTAL flag. + +2010-05-23 - r23255 - lavu 50.16.0 - av_get_random_seed() + Add av_get_random_seed(). + +2010-05-18 - r23161 - lavf 52.63.0 - AVFMT_FLAG_RTP_HINT + Add AVFMT_FLAG_RTP_HINT as possible value for AVFormatContext.flags. + +2010-05-09 - r23066 - lavfi 1.20.0 - AVFilterPicRef + Add interlaced and top_field_first fields to AVFilterPicRef. + +2010-05-01 - r23002 - lavf 52.62.0 - probe function + Add av_probe_input_format2 to API, it allows ignoring probe + results below given score and returns the actual probe score. + +2010-04-01 - r22806 - lavf 52.61.0 - metadata API + Add a flag for av_metadata_set2() to disable overwriting of + existing tags. + +2010-04-01 - r22753 - lavc 52.66.0 + Add avcodec_get_edge_width(). + +2010-03-31 - r22750 - lavc 52.65.0 + Add avcodec_copy_context(). + +2010-03-31 - r22748 - lavf 52.60.0 - av_match_ext() + Make av_match_ext() public. + +2010-03-31 - r22736 - lavu 50.14.0 - AVMediaType + Move AVMediaType enum from libavcodec to libavutil. + +2010-03-31 - r22735 - lavc 52.64.0 - AVMediaType + Define AVMediaType enum, and use it instead of enum CodecType, which + is deprecated and will be dropped at the next major bump. + +2010-03-25 - r22684 - lavu 50.13.0 - av_strerror() + Implement av_strerror(). + +2010-03-23 - r22649 - lavc 52.60.0 - av_dct_init() + Support DCT-I and DST-I. + +2010-03-15 - r22540 - lavf 52.56.0 - AVFormatContext.start_time_realtime + Add AVFormatContext.start_time_realtime field. + +2010-03-13 - r22506 - lavfi 1.18.0 - AVFilterPicRef.pos + Add AVFilterPicRef.pos field. + +2010-03-13 - r22501 - lavu 50.12.0 - error.h + Move error code definitions from libavcodec/avcodec.h to + the new public header libavutil/error.h. + +2010-03-07 - r22291 - lavc 52.56.0 - avfft.h + Add public FFT interface. + +2010-03-06 - r22251 - lavu 50.11.0 - av_stristr() + Add av_stristr(). + +2010-03-03 - r22174 - lavu 50.10.0 - av_tree_enumerate() + Add av_tree_enumerate(). + +2010-02-07 - r21673 - lavu 50.9.0 - av_compare_ts() + Add av_compare_ts(). + +2010-02-05 - r30513 - lsws 0.10.0 - sws_getCoefficients() + Add sws_getCoefficients(). + +2010-02-01 - r21587 - lavf 52.50.0 - metadata API + Add a list of generic tag names, change 'author' -> 'artist', + 'year' -> 'date'. + +2010-01-30 - r21545 - lavu 50.8.0 - av_get_pix_fmt() + Add av_get_pix_fmt(). + +2010-01-21 - r30381 - lsws 0.9.0 - sws_scale() + Change constness attributes of sws_scale() parameters. + +2010-01-10 - r21121 - lavfi 1.15.0 - avfilter_graph_config_links() + Add a log_ctx parameter to avfilter_graph_config_links(). + +2010-01-07 - r30236 - lsws 0.8.0 - sws_isSupported{In,Out}put() + Add sws_isSupportedInput() and sws_isSupportedOutput() functions. + +2010-01-06 - r21035 - lavfi 1.14.0 - avfilter_add_colorspace() + Change the avfilter_add_colorspace() signature, make it accept an + (AVFilterFormats **) rather than an (AVFilterFormats *) as before. + +2010-01-03 - r21007 - lavfi 1.13.0 - avfilter_add_colorspace() + Add avfilter_add_colorspace(). + +2010-01-02 - r20998 - lavf 52.46.0 - av_match_ext() + Add av_match_ext(), it should be used in place of match_ext(). + +2010-01-01 - r20991 - lavf 52.45.0 - av_guess_format() + Add av_guess_format(), it should be used in place of guess_format(). + +2009-12-13 - r20834 - lavf 52.43.0 - metadata API + Add av_metadata_set2(), AV_METADATA_DONT_STRDUP_KEY and + AV_METADATA_DONT_STRDUP_VAL. + +2009-12-13 - r20829 - lavu 50.7.0 - avstring.h API + Add av_d2str(). + +2009-12-13 - r20826 - lavc 52.42.0 - AVStream + Add avg_frame_rate. + +2009-12-12 - r20808 - lavu 50.6.0 - av_bmg_next() + Introduce the av_bmg_next() function. + +2009-12-05 - r20734 - lavfi 1.12.0 - avfilter_draw_slice() + Add a slice_dir parameter to avfilter_draw_slice(). + +2009-11-26 - r20611 - lavfi 1.11.0 - AVFilter + Remove the next field from AVFilter, this is not anymore required. + +2009-11-25 - r20607 - lavfi 1.10.0 - avfilter_next() + Introduce the avfilter_next() function. + +2009-11-25 - r20605 - lavfi 1.9.0 - avfilter_register() + Change the signature of avfilter_register() to make it return an + int. This is required since now the registration operation may fail. + +2009-11-25 - r20603 - lavu 50.5.0 - pixdesc.h API + Make the pixdesc.h API public. + +2009-10-27 - r20385 - lavfi 1.5.0 - AVFilter.next + Add a next field to AVFilter, this is used for simplifying the + registration and management of the registered filters. + +2009-10-23 - r20356 - lavfi 1.4.1 - AVFilter.description + Add a description field to AVFilter. + +2009-10-19 - r20302 - lavfi 1.3.0 - avfilter_make_format_list() + Change the interface of avfilter_make_format_list() from + avfilter_make_format_list(int n, ...) to + avfilter_make_format_list(enum PixelFormat *pix_fmts). + +2009-10-18 - r20272 - lavfi 1.0.0 - avfilter_get_video_buffer() + Make avfilter_get_video_buffer() recursive and add the w and h + parameters to it. + +2009-10-07 - r20189 - lavfi 0.5.1 - AVFilterPic + Add w and h fields to AVFilterPic. + +2009-06-22 - r19250 - lavf 52.34.1 - AVFormatContext.packet_size + This is now an unsigned int instead of a signed int. + +2009-06-19 - r19222 - lavc 52.32.0 - AVSubtitle.pts + Add a pts field to AVSubtitle which gives the subtitle packet pts + in AV_TIME_BASE. Some subtitle de-/encoders (e.g. XSUB) will + not work right without this. + +2009-06-03 - r19078 - lavc 52.30.2 - AV_PKT_FLAG_KEY + PKT_FLAG_KEY has been deprecated and will be dropped at the next + major version. Use AV_PKT_FLAG_KEY instead. + +2009-06-01 - r19025 - lavc 52.30.0 - av_lockmgr_register() + av_lockmgr_register() can be used to register a callback function + that lavc (and in the future, libraries that depend on lavc) can use + to implement mutexes. The application should provide a callback function + that implements the AV_LOCK_* operations described in avcodec.h. + When the lock manager is registered, FFmpeg is guaranteed to behave + correctly in a multi-threaded application. + +2009-04-30 - r18719 - lavc 52.28.0 - av_free_packet() + av_free_packet() is no longer an inline function. It is now exported. + +2009-04-11 - r18431 - lavc 52.25.0 - deprecate av_destruct_packet_nofree() + Please use NULL instead. This has been supported since r16506 + (lavf > 52.23.1, lavc > 52.10.0). + +2009-04-07 - r18351 - lavc 52.23.0 - avcodec_decode_video/audio/subtitle + The old decoding functions are deprecated, all new code should use the + new functions avcodec_decode_video2(), avcodec_decode_audio3() and + avcodec_decode_subtitle2(). These new functions take an AVPacket *pkt + argument instead of a const uint8_t *buf / int buf_size pair. + +2009-04-03 - r18321 - lavu 50.3.0 - av_fifo_space() + Introduce the av_fifo_space() function. + +2009-04-02 - r18317 - lavc 52.23.0 - AVPacket + Move AVPacket declaration from libavformat/avformat.h to + libavcodec/avcodec.h. + +2009-03-22 - r18163 - lavu 50.2.0 - RGB32 pixel formats + Convert the pixel formats PIX_FMT_ARGB, PIX_FMT_RGBA, PIX_FMT_ABGR, + PIX_FMT_BGRA, which were defined as macros, into enum PixelFormat values. + Conversely PIX_FMT_RGB32, PIX_FMT_RGB32_1, PIX_FMT_BGR32 and + PIX_FMT_BGR32_1 are now macros. + avcodec_get_pix_fmt() now recognizes the "rgb32" and "bgr32" aliases. + Re-sort the enum PixelFormat list accordingly. + This change breaks API/ABI backward compatibility. + +2009-03-22 - r18133 - lavu 50.1.0 - PIX_FMT_RGB5X5 endian variants + Add the enum PixelFormat values: + PIX_FMT_RGB565BE, PIX_FMT_RGB565LE, PIX_FMT_RGB555BE, PIX_FMT_RGB555LE, + PIX_FMT_BGR565BE, PIX_FMT_BGR565LE, PIX_FMT_BGR555BE, PIX_FMT_BGR555LE. + +2009-03-21 - r18116 - lavu 50.0.0 - av_random* + The Mersenne Twister PRNG implemented through the av_random* functions + was removed. Use the lagged Fibonacci PRNG through the av_lfg* functions + instead. + +2009-03-08 - r17869 - lavu 50.0.0 - AVFifoBuffer + av_fifo_init, av_fifo_read, av_fifo_write and av_fifo_realloc were dropped + and replaced by av_fifo_alloc, av_fifo_generic_read, av_fifo_generic_write + and av_fifo_realloc2. + In addition, the order of the function arguments of av_fifo_generic_read + was changed to match av_fifo_generic_write. + The AVFifoBuffer/struct AVFifoBuffer may only be used in an opaque way by + applications, they may not use sizeof() or directly access members. + +2009-03-01 - r17682 - lavf 52.31.0 - Generic metadata API + Introduce a new metadata API (see av_metadata_get() and friends). + The old API is now deprecated and should not be used anymore. This especially + includes the following structure fields: + - AVFormatContext.title + - AVFormatContext.author + - AVFormatContext.copyright + - AVFormatContext.comment + - AVFormatContext.album + - AVFormatContext.year + - AVFormatContext.track + - AVFormatContext.genre + - AVStream.language + - AVStream.filename + - AVProgram.provider_name + - AVProgram.name + - AVChapter.title diff --git a/lib/ffmpeg/doc/TODO b/lib/ffmpeg/doc/TODO new file mode 100644 index 0000000000..747eee4ab1 --- /dev/null +++ b/lib/ffmpeg/doc/TODO @@ -0,0 +1,82 @@ +ffmpeg TODO list: +---------------- + +Fabrice's TODO list: (unordered) +------------------- +Short term: + +- 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 + +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. + +Baptiste's TODO list: +----------------- +- mov edit list support (AVEditList) +- YUV 10 bit per component support "2vuy" +- mxf muxer +- mpeg2 non linear quantizer + +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 +- SNOW: try B frames and MCTF and see how their PSNR/bitrate/complexity behaves +- SNOW: try to use the wavelet transformed MC-ed reference frame as context for the entropy coder +- SNOW: think about/analyize how to make snow use multiple cpus/threads +- SNOW: finish spec +- 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) +- regression tests for codecs which do not have an encoder (I+P-frame bitstream in svn) +- add support for using mplayers video filters to ffmpeg +- H264 encoder +- per MB ratecontrol (so VCD and such do work better) +- 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 +- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc +- generic audio mixing API +- extract PES packetizer from PS muxer and use it for new TS muxer +- implement automatic AVBistreamFilter activation +- make cabac encoder use bytestream (see http://trac.videolan.org/x264/changeset/?format=diff&new=651) +- merge imdct and windowing, the current code does considerable amounts of redundant work diff --git a/lib/ffmpeg/doc/avutil.txt b/lib/ffmpeg/doc/avutil.txt new file mode 100644 index 0000000000..210bd07264 --- /dev/null +++ b/lib/ffmpeg/doc/avutil.txt @@ -0,0 +1,37 @@ +AVUtil +====== +libavutil is a small lightweight library of generally useful functions. +It is not a library for code needed by both libavcodec and libavformat. + + +Overview: +========= +adler32.c adler32 checksum +aes.c AES encryption and decryption +fifo.c resizeable first in first out buffer +intfloat_readwrite.c portable reading and writing of floating point values +log.c "printf" with context and level +md5.c MD5 Message-Digest Algorithm +rational.c code to perform exact calculations with rational numbers +tree.c generic AVL tree +crc.c generic CRC checksumming code +integer.c 128bit integer math +lls.c +mathematics.c greatest common divisor, integer sqrt, integer log2, ... +mem.c memory allocation routines with guaranteed alignment +softfloat.c + +Headers: +bswap.h big/little/native-endian conversion code +x86_cpu.h a few useful macros for unifying x86-64 and x86-32 code +avutil.h +common.h +intreadwrite.h reading and writing of unaligned big/little/native-endian integers + + +Goals: +====== +* Modular (few interdependencies and the possibility of disabling individual parts during ./configure) +* Small (source and object) +* Efficient (low CPU and memory usage) +* Useful (avoid useless features almost no one needs) diff --git a/lib/ffmpeg/doc/developer.texi b/lib/ffmpeg/doc/developer.texi new file mode 100644 index 0000000000..edce7ea63a --- /dev/null +++ b/lib/ffmpeg/doc/developer.texi @@ -0,0 +1,436 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Developer Documentation +@titlepage +@sp 7 +@center @titlefont{Developer Documentation} +@sp 3 +@end titlepage + + +@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{libavformat/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. + +@anchor{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 will not +accept patches to remove their use unless they absolutely do not 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 do not 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 one inspired 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. + +The main priority in FFmpeg is simplicity and small code size in order to +minimize the bug count. + +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 is 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. + +Casts should be used only when necessary. Unneeded parentheses +should also be avoided if they don't make the code easier to understand. + +@section Development Policy + +@enumerate +@item + Contributions should be licensed under the LGPL 2.1, including an + "or any later version" clause, or the MIT license. GPL 2 including + an "or any later version" clause is also acceptable, but LGPL is + preferred. +@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 do not 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. Also do not forget that if part B depends on part A, but A does not + depend on B, then A can and should be committed first and separate from B. + Keeping changes well split into self-contained parts makes reviewing and + understanding them on the commit log mailing list easier. This also helps + in case of debugging later on. + Also if you have doubts about splitting or not splitting, do not hesitate to + ask/discuss it on the developer mailing list. +@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 do not.). 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 (do not + 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 + When applying patches that have been discussed (at length) on the mailing + list, reference the thread in the log message. +@item + Do NOT commit to code actively maintained by others without permission. + Send a patch to ffmpeg-devel instead. If no one 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 is 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 + Try to keep important discussions and requests (also) on the public + developer mailing list, so that all developers can benefit from them. +@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. + 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 or extension of an + existing data structure). + Incrementing the third component means a noteworthy binary compatible + change (e.g. encoder bug fix that matters for the decoder). +@item + Compiler warnings indicate potential bugs or code with bad style. If a type of + warning always points to correct and clean code, that warning should + be disabled, not the code changed. + Thus the remaining warnings can either be bugs or correct code. + If it is a bug, the bug has to be fixed. If it is not, the code should + be changed to not generate a warning unless that causes a slowdown + or obfuscates the code. +@item + If you add a new file, give it a proper license header. Do not copy and + paste it from a random place, use an existing file as template. +@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 did not yet. + +When you submit your patch, try to send a unified diff (diff '-up' +option). We cannot read other diffs :-) + +Also please do not submit a patch which contains several unrelated changes. +Split it into separate, self-contained pieces. This does not mean splitting +file by file. Instead, make the patch as small as possible while still +keeping it as a logical unit that contains an individual change, even +if it spans multiple files. This makes reviewing your patches much easier +for us and greatly increases your chances of getting your patch applied. + +Use the patcheck tool of FFmpeg to check your patch. +The tool is located in the tools directory. + +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 will not 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()') + +Also please if you send several patches, send each patch as a separate mail, +do not attach several unrelated patches to the same mail. + +Your patch will be reviewed on the mailing list. You will likely be asked +to make some changes and are expected to send in an improved version that +incorporates the requests from the review. This process may go through +several iterations. Once your patch is deemed good enough, some developer +will pick it up and commit it to the official FFmpeg tree. + +Give us a few days to react. But if some time passes without reaction, +send a reminder by email. Your patch should eventually be dealt with. + + +@section New codecs or formats checklist + +@enumerate +@item + Did you use av_cold for codec initialization and close functions? +@item + Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or + AVInputFormat/AVOutputFormat struct? +@item + Did you bump the minor version number in @file{avcodec.h} or + @file{avformat.h}? +@item + Did you register it in @file{allcodecs.c} or @file{allformats.c}? +@item + Did you add the CodecID to @file{avcodec.h}? +@item + If it has a fourcc, did you add it to @file{libavformat/riff.c}, + even if it is only a decoder? +@item + Did you add a rule to compile the appropriate files in the Makefile? + Remember to do this even if you're just adding a format to a file that is + already being compiled by some other rule, like a raw demuxer. +@item + Did you add an entry to the table of supported formats or codecs in + @file{doc/general.texi}? +@item + Did you add an entry in the Changelog? +@item + If it depends on a parser or a library, did you add that dependency in + configure? +@item + Did you "svn add" the appropriate files before commiting? +@end enumerate + +@section patch submission checklist + +@enumerate +@item + Do the regression tests pass with the patch applied? +@item + Does @code{make checkheaders} pass with the patch applied? +@item + Is the patch a unified diff? +@item + Is the patch against latest FFmpeg SVN? +@item + Are you subscribed to ffmpeg-dev? + (the list is subscribers only due to spam) +@item + Have you checked that the changes are minimal, so that the same cannot be + achieved with a smaller patch and/or simpler final code? +@item + If the change is to speed critical code, did you benchmark it? +@item + If you did any benchmarks, did you provide them in the mail? +@item + Have you checked that the patch does not introduce buffer overflows or + other security issues? +@item + Did you test your decoder or demuxer against damaged data? If no, see + tools/trasher and the noise bitstream filter. Your decoder or demuxer + should not crash or end in a (near) infinite loop when fed damaged data. +@item + Is the patch created from the root of the source tree, so it can be + applied with @code{patch -p0}? +@item + Does the patch not mix functional and cosmetic changes? +@item + Did you add tabs or trailing whitespace to the code? Both are forbidden. +@item + Is the patch attached to the email you send? +@item + Is the mime type of the patch correct? It should be text/x-diff or + text/x-patch or at least text/plain and not application/octet-stream. +@item + If the patch fixes a bug, did you provide a verbose analysis of the bug? +@item + If the patch fixes a bug, did you provide enough information, including + a sample, so the bug can be reproduced and the fix can be verified? + Note please do not attach samples >100k to mails but rather provide a + URL, you can upload to ftp://upload.ffmpeg.org +@item + Did you provide a verbose summary about what the patch does change? +@item + Did you provide a verbose explanation why it changes things like it does? +@item + Did you provide a verbose summary of the user visible advantages and + disadvantages if the patch is applied? +@item + Did you provide an example so we can verify the new feature added by the + patch easily? +@item + If you added a new file, did you insert a license header? It should be + taken from FFmpeg, not randomly copied and pasted from somewhere else. +@item + You should maintain alphabetical order in alphabetically ordered lists as + long as doing so does not break API/ABI compatibility. +@item + Lines with similar content should be aligned vertically when doing so + improves readability. +@item + Did you provide a suggestion for a clear commit log message? +@end enumerate + +@section Patch review process + +All patches posted to ffmpeg-devel will be reviewed, unless they contain a +clear note that the patch is not for SVN. +Reviews and comments will be posted as replies to the patch on the +mailing list. The patch submitter then has to take care of every comment, +that can be by resubmitting a changed patch or by discussion. Resubmitted +patches will themselves be reviewed like any other patch. If at some point +a patch passes review with no comments then it is approved, that can for +simple and small patches happen immediately while large patches will generally +have to be changed and reviewed many times before they are approved. +After a patch is approved it will be committed to the repository. + +We will review all submitted patches, but sometimes we are quite busy so +especially for large patches this can take several weeks. + +When resubmitting patches, please do not make any significant changes +not related to the comments received during review. Such patches will +be rejected. Instead, submit significant changes or new features as +separate patches. + +@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 output is checked immediately after each test +has run. + +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. Commands like +'make regtest-mpeg2' can be used to run a single test. By default, +make will abort if any test fails. To run all tests regardless, +use make -k. To get a more verbose output, use 'make V=1 test' or +'make V=2 test'. + +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/lib/ffmpeg/doc/faq.texi b/lib/ffmpeg/doc/faq.texi new file mode 100644 index 0000000000..3f17738940 --- /dev/null +++ b/lib/ffmpeg/doc/faq.texi @@ -0,0 +1,501 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg FAQ +@titlepage +@sp 7 +@center @titlefont{FFmpeg FAQ} +@sp 3 +@end titlepage + + +@chapter General Questions + +@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 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 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 or sponsor a developer. + +@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it? + +No. Windows DLLs are not portable, bloated and often slow. +Moreover FFmpeg strives to support all codecs natively. +A DLL loader is not conducive to that goal. + +@section My bug report/mail to ffmpeg-devel/user has not received any replies. + +Likely reasons +@itemize +@item We are busy and haven't had time yet to read your report or +investigate the issue. +@item You didn't follow bugreports.html. +@item You didn't use Subversion HEAD. +@item You reported a segmentation fault without gdb output. +@item You describe a problem but not how to reproduce it. +@item It's unclear if you use ffmpeg as command line tool or use +libav* from another application. +@item You speak about a video having problems on playback but +not what you use to play it. +@item We have no faint clue what you are talking about besides +that it is related to FFmpeg. +@end itemize + +@section Is there a forum for FFmpeg? I do not like mailing lists. + +You may view our mailing lists with a more forum-alike look here: +@url{http://dir.gmane.org/gmane.comp.video.ffmpeg.user}, +but, if you post, please remember that our mailing list rules still apply there. + +@section I cannot read this file although this format seems to be supported by ffmpeg. + +Even if ffmpeg can read the container format, it may not support all its +codecs. Please consult the supported codec list in the ffmpeg +documentation. + +@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 like ffdshow or Xvid installed. +@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 +always +@item mp3 +If some MP3 codec like LAME is installed. +@end table + + +@chapter Compilation + +@section @code{error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'} + +This is a bug in gcc. Do not report it to us. Instead, please report it to +the gcc developers. Note that we will not add workarounds for gcc bugs. + +Also note that (some of) the gcc developers believe this is not a bug or +not a bug they should fix: +@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}. +Then again, some of them do not know the difference between an undecidable +problem and an NP-hard problem... + +@chapter Usage + +@section ffmpeg does not work; what is wrong? + +Try a @code{make distclean} in the ffmpeg source directory before the build. If this does not help see +(@url{http://ffmpeg.org/bugreports.html}). + +@section How do I encode single pictures into movies? + +First, rename your pictures to follow a numerical sequence. +For example, img1.jpg, img2.jpg, img3.jpg,... +Then you may run: + +@example + ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg +@end example + +Notice that @samp{%d} is replaced by the image number. + +@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc... + +If you have large number of pictures to rename, you can use the +following command to ease the burden. The command, using the bourne +shell syntax, symbolically links all files in the current directory +that match @code{*jpg} to the @file{/tmp} directory in the sequence of +@file{img001.jpg}, @file{img002.jpg} and so on. + +@example + x=1; for i in *jpg; do counter=$(printf %03d $x); ln "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done +@end example + +If you want to sequence them by oldest modified first, substitute +@code{$(ls -r -t *jpg)} in place of @code{*jpg}. + +Then run: + +@example + ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg +@end example + +The same logic is used for any image format that ffmpeg reads. + +@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 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 file name. + +@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://svn.ffmpeg.org/michael/trunk/patches/bttv-420-2.4.26.patch?view=co}). 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://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.4.26.patch?view=co}) +or (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.6.6.patch?view=co}) +and pass 'combfilter=2'. + +@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 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 in 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 libfaac -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 libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X output.mp4 +@end table + +@section How do I encode videos which play on the PSP? + +@table @option +@item needed stuff +-acodec libfaac -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 libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp output.mp4 +@item needed stuff for H.264 +-acodec libfaac -vcodec libx264 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 libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4 +@item higher resolution for newer PSP firmwares, width<=480, height<=272 +-vcodec libx264 -level 21 -coder 1 -f psp +@item example command line +ffmpeg -i input -acodec libfaac -ab 128kb -ac 2 -ar 48000 -vcodec libx264 -level 21 -b 640kb -coder 1 -f psp -flags +loop -trellis 2 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -s 480x272 output.mp4 +@end table + +@section Which are good parameters for encoding high quality MPEG-4? + +'-mbd rd -flags +4mv+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2', +things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'. + +@section Which are good parameters for encoding high quality MPEG-1/MPEG-2? + +'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2' +but beware the '-g 100' might cause problems with some decoders. +Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd. + +@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong? + +You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced +material, and try '-top 0/1' if the result looks really messed-up. + +@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. + +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/}. + +@section How can I join video files? + +A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by +merely concatenating them. + +Hence you may concatenate your multimedia files by first transcoding them to +these privileged formats, then using the humble @code{cat} command (or the +equally humble @code{copy} under Windows), and finally transcoding back to your +format of choice. + +@example +ffmpeg -i input1.avi -sameq intermediate1.mpg +ffmpeg -i input2.avi -sameq intermediate2.mpg +cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg +ffmpeg -i intermediate_all.mpg -sameq output.avi +@end example + +Notice that you should either use @code{-sameq} or set a reasonably high +bitrate for your intermediate and output files, if you want to preserve +video quality. + +Also notice that you may avoid the huge intermediate files by taking advantage +of named pipes, should your platform support it: + +@example +mkfifo intermediate1.mpg +mkfifo intermediate2.mpg +ffmpeg -i input1.avi -sameq -y intermediate1.mpg < /dev/null & +ffmpeg -i input2.avi -sameq -y intermediate2.mpg < /dev/null & +cat intermediate1.mpg intermediate2.mpg |\ +ffmpeg -f mpeg -i - -sameq -vcodec mpeg4 -acodec libmp3lame output.avi +@end example + +Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also +allow concatenation, and the transcoding step is almost lossless. +When using multiple yuv4mpegpipe(s), the first line needs to be discarded +from all but the first stream. This can be accomplished by piping through +@code{tail} as seen below. Note that when piping through @code{tail} you +must use command grouping, @code{@{ ;@}}, to background properly. + +For example, let's say we want to join two FLV files into an output.flv file: + +@example +mkfifo temp1.a +mkfifo temp1.v +mkfifo temp2.a +mkfifo temp2.v +mkfifo all.a +mkfifo all.v +ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null & +ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null & +ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null & +@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} & +cat temp1.a temp2.a > all.a & +cat temp1.v temp2.v > all.v & +ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \ + -f yuv4mpegpipe -i all.v \ + -sameq -y output.flv +rm temp[12].[av] all.[av] +@end example + +@section FFmpeg does not adhere to the -maxrate setting, some frames are bigger than maxrate/fps. + +Read the MPEG spec about video buffer verifier. + +@section I want CBR, but no matter what I do frame sizes differ. + +You do not understand what CBR is, please read the MPEG spec. +Read about video buffer verifier and constant bitrate. +The one sentence summary is that there is a buffer and the input rate is +constant, the output can vary as needed. + +@section How do I check if a stream is CBR? + +To quote the MPEG-2 spec: +"There is no way to tell that a bitstream is constant bitrate without +examining all of the vbv_delay values and making complicated computations." + + +@chapter Development + +@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.html}). + +@section Can you support my C compiler XXX? + +It depends. If your compiler is C99-compliant, then patches to support +it are likely to be welcome if they do not pollute the source code +with @code{#ifdef}s related to the compiler. + +@section Is Microsoft Visual C++ supported? + +No. Microsoft Visual C++ is not compliant to the C99 standard and does +not - among other things - support the inline assembly used in FFmpeg. +If you wish to use MSVC++ for your +project then you can link the MSVC++ code with libav* as long as +you compile the latter with a working C compiler. For more information, see +the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg +documentation. + +There have been efforts to make FFmpeg compatible with MSVC++ in the +past. However, they have all been rejected as too intrusive, especially +since MinGW does the job adequately. None of the core developers +work with MSVC++ and thus this item is low priority. Should you find +the silver bullet that solves this problem, feel free to shoot it at us. + +We strongly recommend you to move over from MSVC++ to MinGW tools. + +@section Can I use FFmpeg or libavcodec under Windows? + +Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg. +Read the @emph{Windows} section in the FFmpeg documentation to find more +information. + +To get help and instructions for building FFmpeg under Windows, check out +the FFmpeg Windows Help Forum at +@url{http://ffmpeg.arrozcru.org/}. + +@section Can you add automake, libtool or autoconf support? + +No. These tools are too bloated and they complicate the build. + +@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://www.tux.org/lkml/#s15}). + +@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 stripped 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 #if CONFIG_GPL without breaking anything. So for example a new codec +or filter would be OK under GPL while a bug fix 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 I'm using libavcodec from within my C++ application but the linker complains about missing symbols which seem to be available. + +FFmpeg is a pure C project, so to use the libraries within your C++ application +you need to explicitly state that you are using a C library. You can do this by +encompassing your FFmpeg includes using @code{extern "C"}. + +See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3} + +@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. + +@section I get "./configure: line <xxx>: pr: command not found" in MSys. + +The standard MSys install doesn't come with pr. You need to get it from the coreutils package. + +@section I tried to pass RTP packets into a decoder, but it doesn't work. + +RTP is a container format like any other, you must first depacketize the +codec frames/samples stored in RTP and then feed to the decoder. + +@section Where can I find libav* headers for Pascal/Delphi? + +see @url{http://www.iversenit.dk/dev/ffmpeg-headers/} + +@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm? + +see @url{http://svn.ffmpeg.org/michael/trunk/docs/} + +@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec? + +Even if peculiar since it is network oriented, RTP is a container like any +other. You have to @emph{demux} RTP before feeding the payload to libavcodec. +In this specific case please look at RFC 4629 to see how it should be done. + +@section AVStream.r_frame_rate is wrong, it is much larger than the framerate. + +r_frame_rate is NOT the average framerate, it is the smallest framerate +that can accurately represent all timestamps. So no, it is not +wrong if it is larger than the average! +For example, if you have mixed 25 and 30 fps content, then r_frame_rate +will be 150. + +@bye diff --git a/lib/ffmpeg/doc/ffmpeg-doc.texi b/lib/ffmpeg/doc/ffmpeg-doc.texi new file mode 100644 index 0000000000..7e3abadbb8 --- /dev/null +++ b/lib/ffmpeg/doc/ffmpeg-doc.texi @@ -0,0 +1,984 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Documentation +@titlepage +@sp 7 +@center @titlefont{FFmpeg Documentation} +@sp 3 +@end titlepage + +@chapter Synopsis + +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 + +@chapter Description +@c man begin DESCRIPTION + +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. + +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 output file to 24 fps: +@example +ffmpeg -i input.avi -r 24 output.avi +@end example + +* To force the frame rate of the input file (valid for raw formats only) +to 1 fps and the frame rate of the output file to 24 fps: +@example +ffmpeg -r 1 -i input.m2v -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 DESCRIPTION + +@chapter Options +@c man begin OPTIONS + +@include fftools-common-opts.texi + +@section Main options + +@table @option + +@item -f @var{fmt} +Force format. + +@item -i @var{filename} +input file name + +@item -y +Overwrite output files. + +@item -t @var{duration} +Restrict the transcoded/captured video sequence +to the duration specified in seconds. +@code{hh:mm:ss[.xxx]} syntax is also supported. + +@item -fs @var{limit_size} +Set the file size limit. + +@item -ss @var{position} +Seek to given time position in seconds. +@code{hh:mm:ss[.xxx]} syntax is also supported. + +@item -itsoffset @var{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 -timestamp @var{time} +Set the recording timestamp in the container. +The syntax for @var{time} is: +@example +now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.m...]]]))[Z|z]) +@end example +If the value is "now" it takes the current time. +Time is local time unless 'Z' or 'z' is appended, in which case it is +interpreted as UTC. +If the year-month-day part is not specified it takes the current +year-month-day. + +@item -metadata @var{key}=@var{value} +Set a metadata key/value pair. + +For example, for setting the title in the output file: +@example +ffmpeg -i in.avi -metadata title="my title" out.flv +@end example + +@item -v @var{number} +Set the logging verbosity level. + +@item -target @var{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 @var{number} +Set the number of data frames to record. + +@item -scodec @var{codec} +Force subtitle codec ('copy' to copy stream). + +@item -newsubtitle +Add a new subtitle stream to the current output stream. + +@item -slang @var{code} +Set the ISO 639 language code (3 letters) of the current subtitle stream. + +@end table + +@section Video Options + +@table @option +@item -b @var{bitrate} +Set the video bitrate in bit/s (default = 200 kb/s). +@item -vframes @var{number} +Set the number of video frames to record. +@item -r @var{fps} +Set frame rate (Hz value, fraction or abbreviation), (default = 25). +@item -s @var{size} +Set frame size. The format is @samp{wxh} (ffserver default = 160x128, ffmpeg default = same as source). +The following abbreviations are recognized: +@table @samp +@item sqcif +128x96 +@item qcif +176x144 +@item cif +352x288 +@item 4cif +704x576 +@item 16cif +1408x1152 +@item qqvga +160x120 +@item qvga +320x240 +@item vga +640x480 +@item svga +800x600 +@item xga +1024x768 +@item uxga +1600x1200 +@item qxga +2048x1536 +@item sxga +1280x1024 +@item qsxga +2560x2048 +@item hsxga +5120x4096 +@item wvga +852x480 +@item wxga +1366x768 +@item wsxga +1600x1024 +@item wuxga +1920x1200 +@item woxga +2560x1600 +@item wqsxga +3200x2048 +@item wquxga +3840x2400 +@item whsxga +6400x4096 +@item whuxga +7680x4800 +@item cga +320x200 +@item ega +640x350 +@item hd480 +852x480 +@item hd720 +1280x720 +@item hd1080 +1920x1080 +@end table + +@item -aspect @var{aspect} +Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777). +@item -croptop @var{size} (deprecated - use -vf crop=x:y:width:height instead) +Set top crop band size (in pixels). +@item -cropbottom @var{size} (deprecated - use -vf crop=x:y:width:height instead) +Set bottom crop band size (in pixels). +@item -cropleft @var{size} (deprecated - use -vf crop=x:y:width:height instead) +Set left crop band size (in pixels). +@item -cropright @var{size} (deprecated - use -vf crop=x:y:width:height instead) +Set right crop band size (in pixels). +@item -padtop @var{size} +@item -padbottom @var{size} +@item -padleft @var{size} +@item -padright @var{size} +@item -padcolor @var{hex_color} +All the pad options have been removed. Use -vf +pad=width:height:x:y:color instead. +@item -vn +Disable video recording. +@item -bt @var{tolerance} +Set video bitrate tolerance (in bits, default 4000k). +Has a minimum value of: (target_bitrate/target_framerate). +In 1-pass mode, bitrate tolerance specifies how far ratecontrol is +willing to deviate from the target average bitrate value. This is +not related to min/max bitrate. Lowering tolerance too much has +an adverse effect on quality. +@item -maxrate @var{bitrate} +Set max video bitrate (in bit/s). +Requires -bufsize to be set. +@item -minrate @var{bitrate} +Set min video bitrate (in bit/s). +Most useful in setting up a CBR encode: +@example +ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v +@end example +It is of little use elsewise. +@item -bufsize @var{size} +Set video buffer verifier buffer size (in bits). +@item -vcodec @var{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 @var{n} +Select the pass number (1 or 2). It is used to do two-pass +video encoding. The statistics of the video are recorded in the first +pass into a log file (see also the option -passlogfile), +and in the second pass that log file is used to generate the video +at the exact requested bitrate. +On pass 1, you may just deactivate audio and set output to null, +examples for Windows and Unix: +@example +ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL +ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null +@end example + +@item -passlogfile @var{prefix} +Set two-pass log file name prefix to @var{prefix}, the default file name +prefix is ``ffmpeg2pass''. The complete file name will be +@file{PREFIX-N.log}, where N is a number specific to the output +stream. + +@item -newvideo +Add a new video stream to the current output stream. + +@item -vlang @var{code} +Set the ISO 639 language code (3 letters) of the current video stream. + +@item -vf @var{filter_graph} +@var{filter_graph} is a description of the filter graph to apply to +the input video. +Use the option "-filters" to show all the available filters (including +also sources and sinks). + +@end table + +@section Advanced Video Options + +@table @option +@item -pix_fmt @var{format} +Set pixel format. Use 'list' as parameter to show all the supported +pixel formats. +@item -sws_flags @var{flags} +Set SwScaler flags. +@item -g @var{gop_size} +Set the group of pictures size. +@item -intra +Use only intra frames. +@item -vdt @var{n} +Discard threshold. +@item -qscale @var{q} +Use fixed video quantizer scale (VBR). +@item -qmin @var{q} +minimum video quantizer scale (VBR) +@item -qmax @var{q} +maximum video quantizer scale (VBR) +@item -qdiff @var{q} +maximum difference between the quantizer scales (VBR) +@item -qblur @var{blur} +video quantizer scale blur (VBR) (range 0.0 - 1.0) +@item -qcomp @var{compression} +video quantizer scale compression (VBR) (default 0.5). +Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0 + +@item -lmin @var{lambda} +minimum video lagrange factor (VBR) +@item -lmax @var{lambda} +max video lagrange factor (VBR) +@item -mblmin @var{lambda} +minimum macroblock quantizer scale (VBR) +@item -mblmax @var{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 @var{complexity} +initial complexity for single pass encoding +@item -b_qfactor @var{factor} +qp factor between P- and B-frames +@item -i_qfactor @var{factor} +qp factor between P- and I-frames +@item -b_qoffset @var{offset} +qp offset between P- and B-frames +@item -i_qoffset @var{offset} +qp offset between P- and I-frames +@item -rc_eq @var{equation} +Set rate control equation (@pxref{FFmpeg formula +evaluator}) (default = @code{tex^qComp}). +@item -rc_override @var{override} +rate control override for specific intervals +@item -me_method @var{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 hex +@item umh +@item epzs +(default method) +@item full +exhaustive search (slow and marginally better than epzs) +@end table + +@item -dct_algo @var{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 @var{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 @var{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 @var{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 @var{frames} +Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4). +@item -mbd @var{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 @var{param} +Work around encoder bugs that are not auto-detected. +@item -strict @var{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 -vstats_file @var{file} +Dump video coding statistics to @var{file}. +@item -top @var{n} +top=1/bottom=0/auto=-1 field first +@item -dc @var{precision} +Intra_dc_precision. +@item -vtag @var{fourcc/tag} +Force video tag/fourcc. +@item -qphist +Show QP histogram. +@item -vbsf @var{bitstream_filter} +Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump". +@example +ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264 +@end example +@end table + +@section Audio Options + +@table @option +@item -aframes @var{number} +Set the number of audio frames to record. +@item -ar @var{freq} +Set the audio sampling frequency (default = 44100 Hz). +@item -ab @var{bitrate} +Set the audio bitrate in bit/s (default = 64k). +@item -aq @var{q} +Set the audio quality (codec-specific, VBR). +@item -ac @var{channels} +Set the number of audio channels (default = 1). +@item -an +Disable audio recording. +@item -acodec @var{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 384k test.mpg -acodec mp2 -ab 192k -newaudio +@end example +@item -alang @var{code} +Set the ISO 639 language code (3 letters) of the current audio stream. +@end table + +@section Advanced Audio options: + +@table @option +@item -atag @var{fourcc/tag} +Force audio tag/fourcc. +@item -absf @var{bitstream_filter} +Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp". +@end table + +@section Subtitle options: + +@table @option +@item -scodec @var{codec} +Force subtitle codec ('copy' to copy stream). +@item -newsubtitle +Add a new subtitle stream to the current output stream. +@item -slang @var{code} +Set the ISO 639 language code (3 letters) of the current subtitle stream. +@item -sn +Disable subtitle recording. +@item -sbsf @var{bitstream_filter} +Bitstream filters available are "mov2textsub", "text2movsub". +@example +ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt +@end example +@end table + +@section Audio/Video grab options + +@table @option +@item -vc @var{channel} +Set video grab channel (DV1394 only). +@item -tvstd @var{standard} +Set television standard (NTSC, PAL (SECAM)). +@item -isync +Synchronize read on input. +@end table + +@section Advanced options + +@table @option +@item -map @var{input_stream_id}[:@var{sync_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. +@var{sync_stream_id} if specified sets the input stream to sync +against. +@item -map_meta_data @var{outfile}:@var{infile} +Set meta data information of @var{outfile} from @var{infile}. +@item -debug +Print specific debug info. +@item -benchmark +Show benchmarking information at the end of an encode. +Shows CPU time used and maximum memory consumption. +Maximum memory consumption is not supported on all systems, +it will usually display as 0 if not supported. +@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 @var{size} +Set RTP payload size in bytes. +@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 @var{number_of_times} +Repeatedly loop output for formats that support looping such as animated GIF +(0 will loop the output infinitely). +@item -threads @var{count} +Thread count. +@item -vsync @var{parameter} +Video sync method. +0 Each frame is passed with its timestamp from the demuxer to the muxer +1 Frames will be duplicated and dropped to achieve exactly the requested + constant framerate. +2 Frames are passed through with their timestamp or dropped so as to prevent + 2 frames from having the same timestamp +-1 Chooses between 1 and 2 depending on muxer capabilities. This is the default method. + +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 @var{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. +@item -copyts +Copy timestamps from input to output. +@item -shortest +Finish encoding when the shortest input stream ends. +@item -dts_delta_threshold +Timestamp discontinuity delta threshold. +@item -muxdelay @var{seconds} +Set the maximum demux-decode delay. +@item -muxpreload @var{seconds} +Set the initial demux-decode delay. +@item -streamid @var{output-stream-index}:@var{new-value} +Assign a new value to a stream's stream-id field in the next output file. +All stream-id fields are reset to default for each output file. + +For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for +an output mpegts file: +@example +ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts +@end example +@end table + +@section Preset files + +A preset file contains a sequence of @var{option}=@var{value} pairs, +one for each line, specifying a sequence of options which would be +awkward to specify on the command line. Lines starting with the hash +('#') character are ignored and are used to provide comments. Check +the @file{ffpresets} directory in the FFmpeg source tree for examples. + +Preset files are specified with the @code{vpre}, @code{apre}, +@code{spre}, and @code{fpre} options. The @code{fpre} option takes the +filename of the preset instead of a preset name as input and can be +used for any kind of codec. For the @code{vpre}, @code{apre}, and +@code{spre} options, the options specified in a preset file are +applied to the currently selected codec of the same type as the preset +option. + +The argument passed to the @code{vpre}, @code{apre}, and @code{spre} +preset options identifies the preset file to use according to the +following rules: + +First ffmpeg searches for a file named @var{arg}.ffpreset in the +directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in +the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg}) +in that order. For example, if the argument is @code{libx264-max}, it will +search for the file @file{libx264-max.ffpreset}. + +If no such file is found, then ffmpeg will search for a file named +@var{codec_name}-@var{arg}.ffpreset in the above-mentioned +directories, where @var{codec_name} is the name of the codec to which +the preset file options will be applied. For example, if you select +the video codec with @code{-vcodec libx264} and use @code{-vpre max}, +then it will search for the file @file{libx264-max.ffpreset}. + +@anchor{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 statements are available: @code{ld}, @code{st}, +@code{while}. + +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 atan(x) +@item asin(x) +@item acos(x) +@item exp(x) +@item log(x) +@item abs(x) +@item squish(x) +@item gauss(x) +@item mod(x, y) +@item max(x, y) +@item min(x, y) +@item eq(x, y) +@item gte(x, y) +@item gt(x, y) +@item lte(x, y) +@item lt(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 + +@section Protocols + +The file name 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 -protocols' 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 +@c man begin 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 Hz for MPEG audio, 22050 or 11025 for AC-3). + +@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 +@c man end TIPS + +@chapter Examples +@c man begin EXAMPLES + +@section Video and Audio grabbing + +FFmpeg can grab video and audio from devices given that you specify the input +format and device. + +@example +ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /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://linux.bytesex.org/xawtv/}) by Gerd Knorr. You also +have to set the audio recording levels correctly with a +standard mixer. + +@section X11 grabbing + +FFmpeg can grab the X11 display. + +@example +ffmpeg -f x11grab -s cif -i :0.0 /tmp/out.mpg +@end example + +0.0 is display.screen number of your X11 server, same as +the DISPLAY environment variable. + +@example +ffmpeg -f x11grab -s cif -i :0.0+10,20 /tmp/out.mpg +@end example + +0.0 is display.screen number of your X11 server, same as the DISPLAY environment +variable. 10 is the x-offset and 20 the y-offset for the grabbing. + +@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 22050 Hz 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 64k /tmp/a.mp2 -ab 128k /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 libmp3lame -ab 128k 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-libmp3lame} 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}. + +* You can extract images from a video, or create a video from many images: + +For extracting images from a video: +@example +ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg +@end example + +This will extract one video frame per second from the video and will +output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg}, +etc. Images will be rescaled to fit the new WxH values. + +If you want to extract just a limited number of frames, you can use the +above command in combination with the -vframes or -t option, or in +combination with -ss to start extracting from a certain point in time. + +For creating a video from many images: +@example +ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi +@end example + +The syntax @code{foo-%03d.jpeg} specifies to use a decimal number +composed of three digits padded with zeroes to express the sequence +number. It is the same syntax supported by the C printf function, but +only formats accepting a normal integer are suitable. + +* You can put many streams of the same type in the output: + +@example +ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio +@end example + +In addition to the first video and audio streams, the resulting +output file @file{test12.avi} will contain the second video +and the second audio stream found in the input streams list. + +The @code{-newvideo}, @code{-newaudio} and @code{-newsubtitle} +options have to be specified immediately after the name of the output +file to which you want to add them. +@c man end EXAMPLES + +@include filters.texi + +@ignore + +@setfilename ffmpeg +@settitle FFmpeg video converter + +@c man begin SEEALSO +ffplay(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation +@c man end + +@c man begin AUTHORS +The FFmpeg developers +@c man end + +@end ignore + +@bye diff --git a/lib/ffmpeg/doc/ffplay-doc.texi b/lib/ffmpeg/doc/ffplay-doc.texi new file mode 100644 index 0000000000..5e8032fb59 --- /dev/null +++ b/lib/ffmpeg/doc/ffplay-doc.texi @@ -0,0 +1,173 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFplay Documentation +@titlepage +@sp 7 +@center @titlefont{FFplay Documentation} +@sp 3 +@end titlepage + +@chapter Synopsis + +@example +@c man begin SYNOPSIS +ffplay [options] @file{input_file} +@c man end +@end example + +@chapter Description +@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 Options +@c man begin OPTIONS + +@include fftools-common-opts.texi + +@section Main options + +@table @option +@item -x @var{width} +Force displayed width. +@item -y @var{height} +Force displayed height. +@item -s @var{size} +Set frame size (WxH or abbreviation), needed for videos which don't +contain a header with the frame size like raw YUV. +@item -an +Disable audio. +@item -vn +Disable video. +@item -ss @var{pos} +Seek to a given position in seconds. +@item -t @var{duration} +play <duration> seconds of audio/video +@item -bytes +Seek by bytes. +@item -nodisp +Disable graphical display. +@item -f @var{fmt} +Force format. +@item -window_title @var{title} +Set window title (default is the input filename). +@item -loop @var{number} +Loops movie playback <number> times. 0 means forever. +@item -vf @var{filter_graph} +@var{filter_graph} is a description of the filter graph to apply to +the input video. +Use the option "-filters" to show all the available filters (including +also sources and sinks). + +@end table + +@section Advanced options +@table @option +@item -pix_fmt @var{format} +Set pixel format. +@item -stats +Show the stream duration, the codec parameters, the current position in +the stream and the audio/video synchronisation drift. +@item -debug +Print specific debug info. +@item -bug +Work around bugs. +@item -vismv +Visualize motion vectors. +@item -fast +Non-spec-compliant optimizations. +@item -genpts +Generate pts. +@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 @var{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. +@item -threads @var{count} +Set the thread count. +@item -ast @var{audio_stream_number} +Select the desired audio stream number, counting from 0. The number +refers to the list of all the input audio streams. If it is greater +than the number of audio streams minus one, then the last one is +selected, if it is negative the audio playback is disabled. +@item -vst @var{video_stream_number} +Select the desired video stream number, counting from 0. The number +refers to the list of all the input video streams. If it is greater +than the number of video streams minus one, then the last one is +selected, if it is negative the video playback is disabled. +@item -sst @var{subtitle_stream_number} +Select the desired subtitle stream number, counting from 0. The number +refers to the list of all the input subtitle streams. If it is greater +than the number of subtitle streams minus one, then the last one is +selected, if it is negative the subtitle rendering is disabled. +@item -autoexit +Exit when video is done playing. +@item -exitonkeydown +Exit if any key is pressed. +@item -exitonmousedown +Exit if any mouse button is pressed. +@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 t +Cycle subtitle channel. + +@item w +Show audio waves. + +@item left/right +Seek backward/forward 10 seconds. + +@item down/up +Seek backward/forward 1 minute. + +@item mouse click +Seek to percentage in file corresponding to fraction of width. + +@end table + +@c man end + +@include filters.texi + +@ignore + +@setfilename ffplay +@settitle FFplay media player + +@c man begin SEEALSO +ffmpeg(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation +@c man end + +@c man begin AUTHORS +The FFmpeg developers +@c man end + +@end ignore + +@bye diff --git a/lib/ffmpeg/doc/ffprobe-doc.texi b/lib/ffmpeg/doc/ffprobe-doc.texi new file mode 100644 index 0000000000..a1a11c16fd --- /dev/null +++ b/lib/ffmpeg/doc/ffprobe-doc.texi @@ -0,0 +1,123 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFprobe Documentation +@titlepage +@sp 7 +@center @titlefont{FFprobe Documentation} +@sp 3 +@end titlepage + +@chapter Synopsis + +The generic syntax is: + +@example +@c man begin SYNOPSIS +ffprobe [options] [@file{input_file}] +@c man end +@end example + +@chapter Description +@c man begin DESCRIPTION + +FFprobe gathers information from multimedia streams and prints it in +human- and machine-readable fashion. + +For example it can be used to check the format of the container used +by a multimedia stream and the format and type of each media stream +contained in it. + +If a filename is specified in input, ffprobe will try to open and +probe the file content. If the file cannot be opened or recognized as +a multimedia file, a positive exit code is returned. + +FFprobe may be employed both as a standalone application or in +combination with a textual filter, which may perform more +sophisticated processing, e.g. statistical processing or plotting. + +Options are used to list some of the formats supported by ffprobe or +for specifying which information to display, and for setting how +ffprobe will show it. + +FFprobe output is designed to be easily parsable by a textual filter, +and consists of one or more sections of the form: +@example +[SECTION] +key1=val1 +... +keyN=valN +[/SECTION] +@end example + +Metadata tags stored in the container or in the streams are recognized +and printed in the corresponding ``FORMAT'' or ``STREAM'' section, and +are prefixed by the string ``TAG:''. + +@c man end + +@chapter Options +@c man begin OPTIONS + +@include fftools-common-opts.texi + +@section Main options + +@table @option + +@item -convert_tags +Convert the tag names in the format container to the generic FFmpeg tag names. + +@item -f @var{format} +Force format to use. + +@item -unit +Show the unit of the displayed values. + +@item -prefix +Show a SI prefixes of the displayed values. +Unless ``-byte_binary_prefix'' option is used all the prefix +are decimal. + +@item -byte_binary_prefix +Force the use of binary prefixes for byte values. + +@item -sexagesimal +Use sexagesimal format HH:MM:SS.MICROSECONDS for time values. + +@item -pretty +Prettify the format of the displayed values, it corresponds to the +options ``-unit -prefix -byte_binary_prefix -sexagesimal''. + +@item -show_format +Show information about the container format of the input multimedia +stream. + +All the container format information is printed within a section with +name ``FORMAT''. + +@item -show_streams +Show information about each media stream contained in the input +multimedia stream. + +Each media stream information is printed within a dedicated section +with name ``STREAM''. + +@end table +@c man end + +@ignore + +@setfilename ffprobe +@settitle FFprobe media prober + +@c man begin SEEALSO +ffmpeg(1), ffplay(1), ffserver(1) and the FFmpeg HTML documentation +@c man end + +@c man begin AUTHORS +The FFmpeg developers +@c man end + +@end ignore + +@bye diff --git a/lib/ffmpeg/doc/ffserver-doc.texi b/lib/ffmpeg/doc/ffserver-doc.texi new file mode 100644 index 0000000000..77deb85317 --- /dev/null +++ b/lib/ffmpeg/doc/ffserver-doc.texi @@ -0,0 +1,276 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFserver Documentation +@titlepage +@sp 7 +@center @titlefont{FFserver Documentation} +@sp 3 +@end titlepage + +@chapter Synopsys + +The generic syntax is: + +@example +@c man begin SYNOPSIS +ffserver [options] +@c man end +@end example + +@chapter Description +@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). + +FFserver runs in daemon mode by default; that is, it puts itself in +the background and detaches from its TTY, unless it is launched in +debug mode or a NoDaemon option is specified in the configuration +file. + +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. + +@section How does it work? + +FFserver receives prerecorded files or FFM streams from some ffmpeg +instance as input, then streams them over RTP/RTSP/HTTP. + +An ffserver instance will listen on some port as specified in the +configuration file. You can launch one or more instances of ffmpeg and +send one or more FFM streams to the port where ffserver is expecting +to receive them. Alternately, you can make ffserver launch such ffmpeg +instances at startup. + +Input streams are called feeds, and each one is specified by a <Feed> +section in the configuration file. + +For each feed you can have different output streams in various +formats, each one specified by a <Stream> section in the configuration +file. + +@section Status stream + +FFserver supports an HTTP interface which exposes the current status +of the server. + +Simply point your browser to the address of the special status stream +specified in the configuration file. + +For example if you have: +@example +<Stream status.html> +Format status + +# Only allow local people to get the status +ACL allow localhost +ACL allow 192.168.0.0 192.168.255.255 +</Stream> +@end example + +then the server will post a page with the status information when +the special stream @file{status.html} is requested. + +@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 900 MHz 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 +@code{--enable-libmp3lame} 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 where INPUTFILE +is some file which you can decode with ffmpeg: + +@example +./ffserver -f doc/ffserver.conf & +./ffmpeg -i INPUTFILE 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}. +@c man end + +@chapter Options +@c man begin OPTIONS + +@include fftools-common-opts.texi + +@section Main options + +@table @option +@item -f @var{configfile} +Use @file{configfile} instead of @file{/etc/ffserver.conf}. +@item -n +Enable no-launch mode. This option disables all the Launch directives +within the various <Stream> sections. FFserver will not launch any +ffmpeg instance, so you will have to launch them manually. +@item -d +Enable debug mode. This option increases log verbosity, directs log +messages to stdout and causes ffserver to run in the foreground +rather than as a daemon. +@end table +@c man end + +@ignore + +@setfilename ffserver +@settitle FFserver video server + +@c man begin SEEALSO + +ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffmpeg/doc/ffserver.conf} +example and the FFmpeg HTML documentation +@c man end + +@c man begin AUTHORS +The FFmpeg developers +@c man end + +@end ignore + +@bye diff --git a/lib/ffmpeg/doc/ffserver.conf b/lib/ffmpeg/doc/ffserver.conf new file mode 100644 index 0000000000..217117005c --- /dev/null +++ b/lib/ffmpeg/doc/ffserver.conf @@ -0,0 +1,377 @@ +# 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 HTTP connections that can be handled. It has +# to be defined *before* the MaxClients parameter, since it defines the +# MaxClients maximum limit. +MaxHTTPConnections 2000 + +# 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) +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, qqvga, +# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga, +# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720, +# hd1080 +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> + + +# Transcode an incoming live feed to another live feed, +# using libx264 and video presets + +#<Stream live.h264> +#Format rtp +#Feed feed1.ffm +#VideoCodec libx264 +#VideoFrameRate 24 +#VideoBitRate 100 +#VideoSize 480x272 +#AVPresetVideo default +#AVPresetVideo baseline +#AVOptionVideo flags +global_header +# +#AudioCodec libfaac +#AudioBitRate 32 +#AudioChannels 2 +#AudioSampleRate 22050 +#AVOptionAudio flags +global_header +#</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/lib/ffmpeg/doc/fftools-common-opts.texi b/lib/ffmpeg/doc/fftools-common-opts.texi new file mode 100644 index 0000000000..618441e045 --- /dev/null +++ b/lib/ffmpeg/doc/fftools-common-opts.texi @@ -0,0 +1,89 @@ +All the numerical options, if not specified otherwise, accept in input +a string representing a number, which may contain one of the +International System number postfixes, for example 'K', 'M', 'G'. +If 'i' is appended after the postfix, powers of 2 are used instead of +powers of 10. The 'B' postfix multiplies the value for 8, and can be +appended after another postfix or used alone. This allows using for +example 'KB', 'MiB', 'G' and 'B' as postfix. + +Options which do not take arguments are boolean options, and set the +corresponding value to true. They can be set to false by prefixing +with "no" the option name, for example using "-nofoo" in the +commandline will set to false the boolean option with name "foo". + +@section Generic options + +These options are shared amongst the ff* tools. + +@table @option + +@item -L +Show license. + +@item -h, -?, -help, --help +Show help. + +@item -version +Show version. + +@item -formats +Show available formats. + +The fields preceding the format names have the following meanings: +@table @samp +@item D +Decoding available +@item E +Encoding available +@end table + +@item -codecs +Show available codecs. + +The fields preceding the codec names have the following meanings: +@table @samp +@item D +Decoding available +@item E +Encoding available +@item V/A/S +Video/audio/subtitle codec +@item S +Codec supports slices +@item D +Codec supports direct rendering +@item T +Codec can handle input truncated at random locations instead of only at frame boundaries +@end table + +@item -bsfs +Show available bitstream filters. + +@item -protocols +Show available protocols. + +@item -filters +Show available libavfilter filters. + +@item -pix_fmts +Show available pixel formats. + +@item -loglevel @var{loglevel} +Set the logging level used by the library. +@var{loglevel} is a number or a string containing one of the following values: +@table @samp +@item quiet +@item panic +@item fatal +@item error +@item warning +@item info +@item verbose +@item debug +@end table + +By default the program logs to stderr, if coloring is supported by the +terminal, colors are used to mark errors and warnings. Log coloring +can be disabled setting the environment variable @env{NO_COLOR}. + +@end table diff --git a/lib/ffmpeg/doc/filters.texi b/lib/ffmpeg/doc/filters.texi new file mode 100644 index 0000000000..cd4364a786 --- /dev/null +++ b/lib/ffmpeg/doc/filters.texi @@ -0,0 +1,258 @@ +@chapter Video Filters +@c man begin VIDEO FILTERS + +When you configure your FFmpeg build, you can disable any of the +existing filters using --disable-filters. +The configure output will show the video filters included in your +build. + +Below is a description of the currently available video filters. + +@section crop + +Crop the input video to @var{x}:@var{y}:@var{width}:@var{height}. + +@example +./ffmpeg -i in.avi -vf "crop=0:0:0:240" out.avi +@end example + +@var{x} and @var{y} specify the position of the top-left corner of the +output (non-cropped) area. + +The default value of @var{x} and @var{y} is 0. + +The @var{width} and @var{height} parameters specify the width and height +of the output (non-cropped) area. + +A value of 0 is interpreted as the maximum possible size contained in +the area delimited by the top-left corner at position x:y. + +For example the parameters: + +@example +"crop=100:100:0:0" +@end example + +will delimit the rectangle with the top-left corner placed at position +100:100 and the right-bottom corner corresponding to the right-bottom +corner of the input image. + +The default value of @var{width} and @var{height} is 0. + +@section format + +Convert the input video to one of the specified pixel formats. +Libavfilter will try to pick one that is supported for the input to +the next filter. + +The filter accepts a list of pixel format names, separated by ``:'', +for example ``yuv420p:monow:rgb24''. + +The following command: + +@example +./ffmpeg -i in.avi -vf "format=yuv420p" out.avi +@end example + +will convert the input video to the format ``yuv420p''. + +@section noformat + +Force libavfilter not to use any of the specified pixel formats for the +input to the next filter. + +The filter accepts a list of pixel format names, separated by ``:'', +for example ``yuv420p:monow:rgb24''. + +The following command: + +@example +./ffmpeg -i in.avi -vf "noformat=yuv420p, vflip" out.avi +@end example + +will make libavfilter use a format different from ``yuv420p'' for the +input to the vflip filter. + +@section null + +Pass the source unchanged to the output. + +@section pad + +Add paddings to the input image, and places the original input at the +given coordinates @var{x}, @var{y}. + +It accepts the following parameters: +@var{width}:@var{height}:@var{x}:@var{y}:@var{color}. + +Follows the description of the accepted parameters. + +@table @option +@item width, height + +Specify the size of the output image with the paddings added. If the +value for @var{width} or @var{height} is 0, the corresponding input size +is used for the output. + +The default value of @var{width} and @var{height} is 0. + +@item x, y + +Specify the offsets where to place the input image in the padded area +with respect to the top/left border of the output image. + +The default value of @var{x} and @var{y} is 0. + +@item color + +Specify the color of the padded area, it can be the name of a color +(case insensitive match) or a 0xRRGGBB[AA] sequence. + +The default value of @var{color} is ``black''. + +@end table + +@section pixdesctest + +Pixel format descriptor test filter, mainly useful for internal +testing. The output video should be equal to the input video. + +For example: +@example +format=monow, pixdesctest +@end example + +can be used to test the monowhite pixel format descriptor definition. + +@section scale + +Scale the input video to @var{width}:@var{height} and/or convert the image format. + +For example the command: + +@example +./ffmpeg -i in.avi -vf "scale=200:100" out.avi +@end example + +will scale the input video to a size of 200x100. + +If the input image format is different from the format requested by +the next filter, the scale filter will convert the input to the +requested format. + +If the value for @var{width} or @var{height} is 0, the respective input +size is used for the output. + +If the value for @var{width} or @var{height} is -1, the scale filter will +use, for the respective output size, a value that maintains the aspect +ratio of the input image. + +The default value of @var{width} and @var{height} is 0. + +@section slicify + +Pass the images of input video on to next video filter as multiple +slices. + +@example +./ffmpeg -i in.avi -vf "slicify=32" out.avi +@end example + +The filter accepts the slice height as parameter. If the parameter is +not specified it will use the default value of 16. + +Adding this in the beginning of filter chains should make filtering +faster due to better use of the memory cache. + +@section unsharp + +Sharpen or blur the input video. + +It accepts the following parameters: +@var{luma_msize_x}:@var{luma_msize_y}:@var{luma_amount}:@var{chroma_msize_x}:@var{chroma_msize_y}:@var{chroma_amount} + +Negative values for the amount will blur the input video, while positive +values will sharpen. All parameters are optional and default to the +equivalent of the string '5:5:1.0:0:0:0.0'. + +@table @option + +@item luma_msize_x +Set the luma matrix horizontal size. It can be an integer between 3 +and 13, default value is 5. + +@item luma_msize_y +Set the luma matrix vertical size. It can be an integer between 3 +and 13, default value is 5. + +@item luma_amount +Set the luma effect strength. It can be a float number between -2.0 +and 5.0, default value is 1.0. + +@item chroma_msize_x +Set the chroma matrix horizontal size. It can be an integer between 3 +and 13, default value is 0. + +@item chroma_msize_y +Set the chroma matrix vertical size. It can be an integer between 3 +and 13, default value is 0. + +@item luma_amount +Set the chroma effect strength. It can be a float number between -2.0 +and 5.0, default value is 0.0. + +@end table + +@example +# Strong luma sharpen effect parameters +unsharp=7:7:2.5 + +# Strong blur of both luma and chroma parameters +unsharp=7:7:-2:7:7:-2 + +# Use the default values with @command{ffmpeg} +./ffmpeg -i in.avi -vf "unsharp" out.mp4 +@end example + +@section vflip + +Flip the input video vertically. + +@example +./ffmpeg -i in.avi -vf "vflip" out.avi +@end example + +@c man end VIDEO FILTERS + +@chapter Video Sources +@c man begin VIDEO SOURCES + +Below is a description of the currently available video sources. + +@section nullsrc + +Null video source, never return images. It is mainly useful as a +template and to be employed in analysis / debugging tools. + +It accepts as optional parameter a string of the form +@var{width}:@var{height}, where @var{width} and @var{height} specify the size of +the configured source. + +The default values of @var{width} and @var{height} are respectively 352 +and 288 (corresponding to the CIF size format). + +@c man end VIDEO SOURCES + +@chapter Video Sinks +@c man begin VIDEO SINKS + +Below is a description of the currently available video sinks. + +@section nullsink + +Null video sink, do absolutely nothing with the input video. It is +mainly useful as a template and to be employed in analysis / debugging +tools. + +@c man end VIDEO SINKS + diff --git a/lib/ffmpeg/doc/general.texi b/lib/ffmpeg/doc/general.texi new file mode 100644 index 0000000000..0154b79654 --- /dev/null +++ b/lib/ffmpeg/doc/general.texi @@ -0,0 +1,1067 @@ +\input texinfo @c -*- texinfo -*- + +@settitle General Documentation +@titlepage +@sp 7 +@center @titlefont{General Documentation} +@sp 3 +@end titlepage + + +@chapter external libraries + +FFmpeg can be hooked up with a number of external libraries to add support +for more formats. None of them are used by default, their use has to be +explicitly requested by passing the appropriate flags to @file{./configure}. + +@section OpenCORE AMR + +FFmpeg can make use of the OpenCORE libraries for AMR-NB +decoding/encoding and AMR-WB decoding. + +Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for +installing the libraries. Then pass @code{--enable-libopencore-amrnb} and/or +@code{--enable-libopencore-amrwb} to configure to enable the libraries. + +Note that OpenCORE is under the Apache License 2.0 (see +@url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is +incompatible with the LGPL version 2.1 and GPL version 2. You have to +upgrade FFmpeg's license to LGPL version 3 (or if you have enabled +GPL components, GPL version 3) to use it. + + +@chapter Supported File Formats and Codecs + +You can use the @code{-formats} and @code{-codecs} options 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 Name @tab Encoding @tab Decoding @tab Comments +@item 4xm @tab @tab X + @tab 4X Technologies format, used in some games. +@item 8088flex TMV @tab @tab X +@item Adobe Filmstrip @tab X @tab X +@item Audio IFF (AIFF) @tab X @tab X +@item American Laser Games MM @tab @tab X + @tab Multimedia format used in games like Mad Dog McCree. +@item 3GPP AMR @tab X @tab X +@item ASF @tab X @tab X +@item AVI @tab X @tab X +@item AVISynth @tab @tab X +@item AVS @tab @tab X + @tab Multimedia format used by the Creature Shock game. +@item Beam Software SIFF @tab @tab X + @tab Audio and video format used in some games by Beam Software. +@item Bethesda Softworks VID @tab @tab X + @tab Used in some games from Bethesda Softworks. +@item Bink @tab @tab X + @tab Multimedia format used by many games. +@item Brute Force & Ignorance @tab @tab X + @tab Used in the game Flash Traffic: City of Angels. +@item Interplay C93 @tab @tab X + @tab Used in the game Cyberia from Interplay. +@item Delphine Software International CIN @tab @tab X + @tab Multimedia format used by Delphine Software games. +@item CD+G @tab @tab X + @tab Video format used by CD+G karaoke disks +@item Core Audio Format @tab @tab X + @tab Apple Core Audio Format +@item CRC testing format @tab X @tab +@item Creative Voice @tab X @tab X + @tab Created for the Sound Blaster Pro. +@item CRYO APC @tab @tab X + @tab Audio format used in some games by CRYO Interactive Entertainment. +@item D-Cinema audio @tab X @tab X +@item Deluxe Paint Animation @tab @tab X +@item DV video @tab X @tab X +@item DXA @tab @tab X + @tab This format is used in the non-Windows version of the Feeble Files + game and different game cutscenes repacked for use with ScummVM. +@item Electronic Arts cdata @tab @tab X +@item Electronic Arts Multimedia @tab @tab X + @tab Used in various EA games; files have extensions like WVE and UV2. +@item FFM (FFserver live feed) @tab X @tab X +@item Flash (SWF) @tab X @tab X +@item Flash 9 (AVM2) @tab X @tab X + @tab Only embedded audio is decoded. +@item FLI/FLC/FLX animation @tab @tab X + @tab .fli/.flc files +@item Flash Video (FLV) @tab @tab X + @tab Macromedia Flash video files +@item framecrc testing format @tab X @tab +@item FunCom ISS @tab @tab X + @tab Audio format used in various games from FunCom like The Longest Journey. +@item GIF Animation @tab X @tab +@item GXF @tab X @tab X + @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley + playout servers. +@item id Quake II CIN video @tab @tab X +@item id RoQ @tab X @tab X + @tab Used in Quake III, Jedi Knight 2, other computer games. +@item IEC61937 encapsulation @tab X @tab X +@item IFF @tab @tab X + @tab Interchange File Format +@item Interplay MVE @tab @tab X + @tab Format used in various Interplay computer games. +@item IV8 @tab @tab X + @tab A format generated by IndigoVision 8000 video server. +@item IVF (On2) @tab @tab X + @tab A format used by libvpx +@item LMLM4 @tab @tab X + @tab Used by Linux Media Labs MPEG-4 PCI boards +@item Matroska @tab X @tab X +@item Matroska audio @tab X @tab +@item MAXIS XA @tab @tab X + @tab Used in Sim City 3000; file extension .xa. +@item MD Studio @tab @tab X +@item Monkey's Audio @tab @tab X +@item Motion Pixels MVI @tab @tab X +@item MOV/QuickTime/MP4 @tab X @tab X + @tab 3GP, 3GP2, PSP, iPod variants supported +@item MP2 @tab X @tab X +@item MP3 @tab X @tab X +@item MPEG-1 System @tab X @tab X + @tab muxed audio and video, VCD format supported +@item MPEG-PS (program stream) @tab X @tab X + @tab also known as @code{VOB} file, SVCD and DVD format supported +@item MPEG-TS (transport stream) @tab X @tab X + @tab also known as DVB Transport Stream +@item MPEG-4 @tab X @tab X + @tab MPEG-4 is a variant of QuickTime. +@item MIME multipart JPEG @tab X @tab +@item MSN TCP webcam @tab @tab X + @tab Used by MSN Messenger webcam streams. +@item MTV @tab @tab X +@item Musepack @tab @tab X +@item Musepack SV8 @tab @tab X +@item Material eXchange Format (MXF) @tab X @tab X + @tab SMPTE 377M, used by D-Cinema, broadcast industry. +@item Material eXchange Format (MXF), D-10 Mapping @tab X @tab X + @tab SMPTE 386M, D-10/IMX Mapping. +@item NC camera feed @tab @tab X + @tab NC (AVIP NC4600) camera streams +@item NTT TwinVQ (VQF) @tab @tab X + @tab Nippon Telegraph and Telephone Corporation TwinVQ. +@item Nullsoft Streaming Video @tab @tab X +@item NuppelVideo @tab @tab X +@item NUT @tab X @tab X + @tab NUT Open Container Format +@item Ogg @tab X @tab X +@item TechnoTrend PVA @tab @tab X + @tab Used by TechnoTrend DVB PCI boards. +@item QCP @tab @tab X +@item raw ADTS (AAC) @tab X @tab X +@item raw AC-3 @tab X @tab X +@item raw Chinese AVS video @tab @tab X +@item raw CRI ADX @tab X @tab X +@item raw Dirac @tab X @tab X +@item raw DNxHD @tab X @tab X +@item raw DTS @tab X @tab X +@item raw E-AC-3 @tab X @tab X +@item raw FLAC @tab X @tab X +@item raw GSM @tab @tab X +@item raw H.261 @tab X @tab X +@item raw H.263 @tab X @tab X +@item raw H.264 @tab X @tab X +@item raw Ingenient MJPEG @tab @tab X +@item raw MJPEG @tab X @tab X +@item raw MLP @tab @tab X +@item raw MPEG @tab @tab X +@item raw MPEG-1 @tab @tab X +@item raw MPEG-2 @tab @tab X +@item raw MPEG-4 @tab X @tab X +@item raw NULL @tab X @tab +@item raw video @tab X @tab X +@item raw id RoQ @tab X @tab +@item raw Shorten @tab @tab X +@item raw TrueHD @tab X @tab X +@item raw VC-1 @tab @tab X +@item raw PCM A-law @tab X @tab X +@item raw PCM mu-law @tab X @tab X +@item raw PCM signed 8 bit @tab X @tab X +@item raw PCM signed 16 bit big-endian @tab X @tab X +@item raw PCM signed 16 bit little-endian @tab X @tab X +@item raw PCM signed 24 bit big-endian @tab X @tab X +@item raw PCM signed 24 bit little-endian @tab X @tab X +@item raw PCM signed 32 bit big-endian @tab X @tab X +@item raw PCM signed 32 bit little-endian @tab X @tab X +@item raw PCM unsigned 8 bit @tab X @tab X +@item raw PCM unsigned 16 bit big-endian @tab X @tab X +@item raw PCM unsigned 16 bit little-endian @tab X @tab X +@item raw PCM unsigned 24 bit big-endian @tab X @tab X +@item raw PCM unsigned 24 bit little-endian @tab X @tab X +@item raw PCM unsigned 32 bit big-endian @tab X @tab X +@item raw PCM unsigned 32 bit little-endian @tab X @tab X +@item raw PCM floating-point 32 bit big-endian @tab X @tab X +@item raw PCM floating-point 32 bit little-endian @tab X @tab X +@item raw PCM floating-point 64 bit big-endian @tab X @tab X +@item raw PCM floating-point 64 bit little-endian @tab X @tab X +@item RDT @tab @tab X +@item REDCODE R3D @tab @tab X + @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio. +@item RealMedia @tab X @tab X +@item Redirector @tab @tab X +@item Renderware TeXture Dictionary @tab @tab X +@item RL2 @tab @tab X + @tab Audio and video format used in some games by Entertainment Software Partners. +@item RPL/ARMovie @tab @tab X +@item RTMP @tab X @tab X + @tab Output is performed by publishing stream to RTMP server +@item RTP @tab @tab X +@item RTSP @tab X @tab X +@item SDP @tab @tab X +@item Sega FILM/CPK @tab @tab X + @tab Used in many Sega Saturn console games. +@item Sierra SOL @tab @tab X + @tab .sol files used in Sierra Online games. +@item Sierra VMD @tab @tab X + @tab Used in Sierra CD-ROM games. +@item Smacker @tab @tab X + @tab Multimedia format used by many games. +@item Sony OpenMG (OMA) @tab @tab X + @tab Audio format used in Sony Sonic Stage and Sony Vegas. +@item Sony PlayStation STR @tab @tab X +@item Sony Wave64 (W64) @tab @tab X +@item SoX native format @tab X @tab X +@item SUN AU format @tab X @tab X +@item THP @tab @tab X + @tab Used on the Nintendo GameCube. +@item Tiertex Limited SEQ @tab @tab X + @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback. +@item True Audio @tab @tab X +@item VC-1 test bitstream @tab X @tab X +@item WAV @tab X @tab X +@item WavPack @tab @tab X +@item WebM @tab X @tab X +@item Wing Commander III movie @tab @tab X + @tab Multimedia format used in Origin's Wing Commander III computer game. +@item Westwood Studios audio @tab @tab X + @tab Multimedia format used in Westwood Studios games. +@item Westwood Studios VQA @tab @tab X + @tab Multimedia format used in Westwood Studios games. +@item YUV4MPEG pipe @tab X @tab X +@item Psygnosis YOP @tab @tab X +@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 Name @tab Encoding @tab Decoding @tab Comments +@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 BMP @tab X @tab X + @tab Microsoft BMP image +@item DPX @tab @tab X + @tab Digital Picture Exchange +@item JPEG @tab X @tab X + @tab Progressive JPEG is not supported. +@item JPEG 2000 @tab @tab E + @tab decoding supported through external library libopenjpeg +@item JPEG-LS @tab X @tab X +@item LJPEG @tab X @tab + @tab Lossless JPEG +@item PAM @tab X @tab X + @tab PAM is a PNM extension with alpha support. +@item PBM @tab X @tab X + @tab Portable BitMap image +@item PCX @tab X @tab X + @tab PC Paintbrush +@item PGM @tab X @tab X + @tab Portable GrayMap image +@item PGMYUV @tab X @tab X + @tab PGM with U and V components in YUV 4:2:0 +@item PIC @tab @tab X + @tab Pictor/PC Paint +@item PNG @tab X @tab X + @tab 2/4 bpp not supported yet +@item PPM @tab X @tab X + @tab Portable PixelMap image +@item PTX @tab @tab X + @tab V.Flash PTX format +@item SGI @tab X @tab X + @tab SGI RGB image format +@item Sun Rasterfile @tab @tab X + @tab Sun RAS image format +@item TIFF @tab X @tab X + @tab YUV, JPEG and some extension is not supported yet. +@item Truevision Targa @tab X @tab X + @tab Targa (.TGA) image format +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@code{E} means that support is provided through an external library. + +@section Video Codecs + +@multitable @columnfractions .4 .1 .1 .4 +@item Name @tab Encoding @tab Decoding @tab Comments +@item 4X Movie @tab @tab X + @tab Used in certain computer games. +@item 8088flex TMV @tab @tab X +@item 8SVX exponential @tab @tab X +@item 8SVX fibonacci @tab @tab X +@item American Laser Games MM @tab @tab X + @tab Used in games like Mad Dog McCree. +@item AMV Video @tab @tab X + @tab Used in Chinese MP3 players. +@item Apple MJPEG-B @tab @tab X +@item Apple QuickDraw @tab @tab X + @tab fourcc: qdrw +@item Asus v1 @tab X @tab X + @tab fourcc: ASV1 +@item Asus v2 @tab X @tab X + @tab fourcc: ASV2 +@item ATI VCR1 @tab @tab X + @tab fourcc: VCR1 +@item ATI VCR2 @tab @tab X + @tab fourcc: VCR2 +@item Auravision Aura @tab @tab X +@item Auravision Aura 2 @tab @tab X +@item Autodesk Animator Flic video @tab @tab X +@item Autodesk RLE @tab @tab X + @tab fourcc: AASC +@item AVS (Audio Video Standard) video @tab @tab X + @tab Video encoding used by the Creature Shock game. +@item Beam Software VB @tab @tab X +@item Bethesda VID video @tab @tab X + @tab Used in some games from Bethesda Softworks. +@item Bink Video @tab @tab X + @tab Support for version 'b' is missing. +@item Brute Force & Ignorance @tab @tab X + @tab Used in the game Flash Traffic: City of Angels. +@item C93 video @tab @tab X + @tab Codec used in Cyberia game. +@item CamStudio @tab @tab X + @tab fourcc: CSCD +@item CD+G @tab @tab X + @tab Video codec for CD+G karaoke disks +@item Chinese AVS video @tab @tab X + @tab AVS1-P2, JiZhun profile +@item Delphine Software International CIN video @tab @tab X + @tab Codec used in Delphine Software International games. +@item Cinepak @tab @tab X +@item Cirrus Logic AccuPak @tab @tab X + @tab fourcc: CLJR +@item Creative YUV (CYUV) @tab @tab X +@item Dirac @tab E @tab E + @tab supported through external libdirac/libschroedinger libraries +@item Deluxe Paint Animation @tab @tab X +@item DNxHD @tab X @tab X + @tab aka SMPTE VC3 +@item Duck TrueMotion 1.0 @tab @tab X + @tab fourcc: DUCK +@item Duck TrueMotion 2.0 @tab @tab X + @tab fourcc: TM20 +@item DV (Digital Video) @tab X @tab X +@item Feeble Files/ScummVM DXA @tab @tab X + @tab Codec originally used in Feeble Files game. +@item Electronic Arts CMV video @tab @tab X + @tab Used in NHL 95 game. +@item Electronic Arts Madcow video @tab @tab X +@item Electronic Arts TGV video @tab @tab X +@item Electronic Arts TGQ video @tab @tab X +@item Electronic Arts TQI video @tab @tab X +@item Escape 124 @tab @tab X +@item FFmpeg video codec #1 @tab X @tab X + @tab experimental lossless codec (fourcc: FFV1) +@item Flash Screen Video v1 @tab X @tab X + @tab fourcc: FSV1 +@item Flash Video (FLV) @tab X @tab X + @tab Sorenson H.263 used in Flash +@item Fraps @tab @tab X +@item H.261 @tab X @tab X +@item H.263 / H.263-1996 @tab X @tab X +@item H.263+ / H.263-1998 / H.263 version 2 @tab X @tab X +@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 @tab E @tab X + @tab encoding supported through external library libx264 +@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration) @tab E @tab X +@item HuffYUV @tab X @tab X +@item HuffYUV FFmpeg variant @tab X @tab X +@item IBM Ultimotion @tab @tab X + @tab fourcc: ULTI +@item id Cinematic video @tab @tab X + @tab Used in Quake II. +@item id RoQ video @tab X @tab X + @tab Used in Quake III, Jedi Knight 2, other computer games. +@item IFF ILBM @tab @tab X + @tab IFF interlaved bitmap +@item IFF ByteRun1 @tab @tab X + @tab IFF run length encoded bitmap +@item Intel H.263 @tab @tab X +@item Intel Indeo 2 @tab @tab X +@item Intel Indeo 3 @tab @tab X +@item Intel Indeo 5 @tab @tab X +@item Interplay C93 @tab @tab X + @tab Used in the game Cyberia from Interplay. +@item Interplay MVE video @tab @tab X + @tab Used in Interplay .MVE files. +@item Karl Morton's video codec @tab @tab X + @tab Codec used in Worms games. +@item Kega Game Video (KGV1) @tab @tab X + @tab Kega emulator screen capture codec. +@item LCL (LossLess Codec Library) MSZH @tab @tab X +@item LCL (LossLess Codec Library) ZLIB @tab E @tab E +@item LOCO @tab @tab X +@item lossless MJPEG @tab X @tab X +@item Microsoft RLE @tab @tab X +@item Microsoft Video 1 @tab @tab X +@item Mimic @tab @tab X + @tab Used in MSN Messenger Webcam streams. +@item Miro VideoXL @tab @tab X + @tab fourcc: VIXL +@item MJPEG (Motion JPEG) @tab X @tab X +@item Motion Pixels video @tab @tab X +@item MPEG-1 video @tab X @tab X +@item MPEG-1/2 video XvMC (X-Video Motion Compensation) @tab @tab X +@item MPEG-1/2 video (VDPAU acceleration) @tab @tab X +@item MPEG-2 video @tab X @tab X +@item MPEG-4 part 2 @tab X @tab X + @ libxvidcore can be used alternatively for encoding. +@item MPEG-4 part 2 Microsoft variant version 1 @tab X @tab X +@item MPEG-4 part 2 Microsoft variant version 2 @tab X @tab X +@item MPEG-4 part 2 Microsoft variant version 3 @tab X @tab X +@item Nintendo Gamecube THP video @tab @tab X +@item NuppelVideo/RTjpeg @tab @tab X + @tab Video encoding used in NuppelVideo files. +@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: VP60,VP61,VP62 +@item VP8 @tab E @tab X + @tab fourcc: VP80, encoding supported through external library libvpx +@item planar RGB @tab @tab X + @tab fourcc: 8BPS +@item Q-team QPEG @tab @tab X + @tab fourccs: QPEG, Q1.0, Q1.1 +@item QuickTime 8BPS video @tab @tab X +@item QuickTime Animation (RLE) video @tab X @tab X + @tab fourcc: 'rle ' +@item QuickTime Graphics (SMC) @tab @tab X + @tab fourcc: 'smc ' +@item QuickTime video (RPZA) @tab @tab X + @tab fourcc: rpza +@item R210 Quicktime Uncompressed RGB 10-bit @tab @tab X +@item Raw Video @tab X @tab X +@item RealVideo 1.0 @tab X @tab X +@item RealVideo 2.0 @tab X @tab X +@item RealVideo 3.0 @tab @tab X + @tab still far from ideal +@item RealVideo 4.0 @tab @tab X +@item Renderware TXD (TeXture Dictionary) @tab @tab X + @tab Texture dictionaries used by the Renderware Engine. +@item RL2 video @tab @tab X + @tab used in some games by Entertainment Software Partners +@item Sierra VMD video @tab @tab X + @tab Used in Sierra VMD files. +@item Smacker video @tab @tab X + @tab Video encoding used in Smacker. +@item SMPTE VC-1 @tab @tab X +@item Snow @tab X @tab X + @tab experimental wavelet codec (fourcc: SNOW) +@item Sony PlayStation MDEC (Motion DECoder) @tab @tab X +@item Sorenson Vector Quantizer 1 @tab X @tab X + @tab fourcc: SVQ1 +@item Sorenson Vector Quantizer 3 @tab @tab X + @tab fourcc: SVQ3 +@item Sunplus JPEG (SP5X) @tab @tab X + @tab fourcc: SP5X +@item TechSmith Screen Capture Codec @tab @tab X + @tab fourcc: TSCC +@item Theora @tab E @tab X + @tab encoding supported through external library libtheora +@item Tiertex Limited SEQ video @tab @tab X + @tab Codec used in DOS CD-ROM FlashBack game. +@item V210 Quicktime Uncompressed 4:2:2 10-bit @tab X @tab X +@item VMware Screen Codec / VMware Video @tab @tab X + @tab Codec used in videos captured by VMware. +@item Westwood Studios VQA (Vector Quantized Animation) video @tab @tab X +@item Windows Media Video 7 @tab X @tab X +@item Windows Media Video 8 @tab X @tab X +@item Windows Media Video 9 @tab @tab X + @tab not completely working +@item Wing Commander III / Xan @tab @tab X + @tab Used in Wing Commander III .MVE files. +@item Winnov WNV1 @tab @tab X +@item WMV7 @tab X @tab X +@item YAMAHA SMAF @tab X @tab X +@item Psygnosis YOP Video @tab @tab X +@item ZLIB @tab X @tab X + @tab part of LCL, encoder experimental +@item Zip Motion Blocks Video @tab X @tab X + @tab Encoder works only in PAL8. +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@code{E} means that support is provided through an external library. + +@section Audio Codecs + +@multitable @columnfractions .4 .1 .1 .4 +@item Name @tab Encoding @tab Decoding @tab Comments +@item 8SVX audio @tab @tab X +@item AAC @tab E @tab X + @tab encoding supported through external library libfaac +@item AC-3 @tab IX @tab X +@item ADPCM 4X Movie @tab @tab X +@item ADPCM CDROM XA @tab @tab X +@item ADPCM Creative Technology @tab @tab X + @tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2 +@item ADPCM Electronic Arts @tab @tab X + @tab Used in various EA titles. +@item ADPCM Electronic Arts Maxis CDROM XS @tab @tab X + @tab Used in Sim City 3000. +@item ADPCM Electronic Arts R1 @tab @tab X +@item ADPCM Electronic Arts R2 @tab @tab X +@item ADPCM Electronic Arts R3 @tab @tab X +@item ADPCM Electronic Arts XAS @tab @tab X +@item ADPCM G.726 @tab X @tab X +@item ADPCM IMA AMV @tab @tab X + @tab Used in AMV files +@item ADPCM IMA Electronic Arts EACS @tab @tab X +@item ADPCM IMA Electronic Arts SEAD @tab @tab X +@item ADPCM IMA Funcom @tab @tab X +@item ADPCM IMA QuickTime @tab X @tab X +@item ADPCM IMA Loki SDL MJPEG @tab @tab X +@item ADPCM IMA WAV @tab X @tab X +@item ADPCM IMA Westwood @tab @tab X +@item ADPCM ISS IMA @tab @tab X + @tab Used in FunCom games. +@item ADPCM IMA Duck DK3 @tab @tab X + @tab Used in some Sega Saturn console games. +@item ADPCM IMA Duck DK4 @tab @tab X + @tab Used in some Sega Saturn console games. +@item ADPCM Microsoft @tab X @tab X +@item ADPCM MS IMA @tab X @tab X +@item ADPCM Nintendo Gamecube THP @tab @tab X +@item ADPCM QT IMA @tab X @tab X +@item ADPCM SEGA CRI ADX @tab X @tab X + @tab Used in Sega Dreamcast games. +@item ADPCM Shockwave Flash @tab X @tab X +@item ADPCM SMJPEG IMA @tab @tab X + @tab Used in certain Loki game ports. +@item ADPCM Sound Blaster Pro 2-bit @tab @tab X +@item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X +@item ADPCM Sound Blaster Pro 4-bit @tab @tab X +@item ADPCM Westwood Studios IMA @tab @tab X + @tab Used in Westwood Studios games like Command and Conquer. +@item ADPCM Yamaha @tab X @tab X +@item AMR-NB @tab E @tab X + @tab encoding supported through external library libopencore-amrnb +@item AMR-WB @tab @tab E + @tab decoding supported through external library libopencore-amrwb +@item Apple lossless audio @tab X @tab X + @tab QuickTime fourcc 'alac' +@item Atrac 1 @tab @tab X +@item Atrac 3 @tab @tab X +@item Bink Audio @tab @tab X + @tab Used in Bink and Smacker files in many games. +@item Delphine Software International CIN audio @tab @tab X + @tab Codec used in Delphine Software International games. +@item COOK @tab @tab X + @tab All versions except 5.1 are supported. +@item DCA (DTS Coherent Acoustics) @tab @tab X +@item DPCM id RoQ @tab X @tab X + @tab Used in Quake III, Jedi Knight 2, other computer games. +@item DPCM Interplay @tab @tab X + @tab Used in various Interplay computer games. +@item DPCM Sierra Online @tab @tab X + @tab Used in Sierra Online game audio files. +@item DPCM Sol @tab @tab X +@item DPCM Xan @tab @tab X + @tab Used in Origin's Wing Commander IV AVI files. +@item DSP Group TrueSpeech @tab @tab X +@item DV audio @tab @tab X +@item Enhanced AC-3 @tab @tab X +@item FLAC (Free Lossless Audio Codec) @tab X @tab IX +@item G.729 @tab @tab X +@item GSM @tab E @tab X + @tab encoding supported through external library libgsm +@item GSM Microsoft variant @tab E @tab X + @tab encoding supported through external library libgsm +@item IMC (Intel Music Coder) @tab @tab X +@item MACE (Macintosh Audio Compression/Expansion) 3:1 @tab @tab X +@item MACE (Macintosh Audio Compression/Expansion) 6:1 @tab @tab X +@item MLP (Meridian Lossless Packing) @tab @tab X + @tab Used in DVD-Audio discs. +@item Monkey's Audio @tab @tab X + @tab Only versions 3.97-3.99 are supported. +@item MP1 (MPEG audio layer 1) @tab @tab IX +@item MP2 (MPEG audio layer 2) @tab IX @tab IX +@item MP3 (MPEG audio layer 3) @tab E @tab IX + @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported +@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X +@item Musepack SV7 @tab @tab X +@item Musepack SV8 @tab @tab X +@item Nellymoser Asao @tab X @tab X +@item PCM A-law @tab X @tab X +@item PCM mu-law @tab X @tab X +@item PCM 16-bit little-endian planar @tab @tab X +@item PCM 32-bit floating point big-endian @tab X @tab X +@item PCM 32-bit floating point little-endian @tab X @tab X +@item PCM 64-bit floating point big-endian @tab X @tab X +@item PCM 64-bit floating point little-endian @tab X @tab X +@item PCM D-Cinema audio signed 24-bit @tab X @tab X +@item PCM signed 8-bit @tab X @tab X +@item PCM signed 16-bit big-endian @tab X @tab X +@item PCM signed 16-bit little-endian @tab X @tab X +@item PCM signed 24-bit big-endian @tab X @tab X +@item PCM signed 24-bit little-endian @tab X @tab X +@item PCM signed 32-bit big-endian @tab X @tab X +@item PCM signed 32-bit little-endian @tab X @tab X +@item PCM signed 16/20/24-bit big-endian in MPEG-TS @tab @tab X +@item PCM unsigned 8-bit @tab X @tab X +@item PCM unsigned 16-bit big-endian @tab X @tab X +@item PCM unsigned 16-bit little-endian @tab X @tab X +@item PCM unsigned 24-bit big-endian @tab X @tab X +@item PCM unsigned 24-bit little-endian @tab X @tab X +@item PCM unsigned 32-bit big-endian @tab X @tab X +@item PCM unsigned 32-bit little-endian @tab X @tab X +@item PCM Zork @tab X @tab X +@item QCELP / PureVoice @tab @tab X +@item QDesign Music Codec 2 @tab @tab X + @tab There are still some distortions. +@item RealAudio 1.0 (14.4K) @tab X @tab X + @tab Real 14400 bit/s codec +@item RealAudio 2.0 (28.8K) @tab @tab X + @tab Real 28800 bit/s codec +@item RealAudio 3.0 (dnet) @tab IX @tab X + @tab Real low bitrate AC-3 codec +@item RealAudio SIPR / ACELP.NET @tab @tab X +@item Shorten @tab @tab X +@item Sierra VMD audio @tab @tab X + @tab Used in Sierra VMD files. +@item Smacker audio @tab @tab X +@item Sonic @tab X @tab X + @tab experimental codec +@item Sonic lossless @tab X @tab X + @tab experimental codec +@item Speex @tab @tab E + @tab supported through external library libspeex +@item True Audio (TTA) @tab @tab X +@item TrueHD @tab @tab X + @tab Used in HD-DVD and Blu-Ray discs. +@item TwinVQ (VQF flavor) @tab @tab X +@item Vorbis @tab E @tab X + @tab A native but very primitive encoder exists. +@item WavPack @tab @tab X +@item Westwood Audio (SND1) @tab @tab X +@item Windows Media Audio 1 @tab X @tab X +@item Windows Media Audio 2 @tab X @tab X +@item Windows Media Audio Pro @tab @tab X +@item Windows Media Audio Voice @tab @tab X +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@code{E} means that support is provided through an external library. + +@code{I} means that an integer-only version is available, too (ensures high +performance on systems without hardware floating point support). + +@section Subtitle Formats + +@multitable @columnfractions .4 .1 .1 .1 .1 +@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding +@item SSA/ASS @tab X @tab X +@item DVB @tab X @tab X @tab X @tab X +@item DVD @tab X @tab X @tab X @tab X +@item PGS @tab @tab @tab @tab X +@item XSUB @tab @tab @tab X @tab X +@end multitable + +@code{X} means that the feature is supported. + +@section Network Protocols + +@multitable @columnfractions .4 .1 +@item Name @tab Support +@item file @tab X +@item Gopher @tab X +@item HTTP @tab X +@item MMS @tab X +@item pipe @tab X +@item RTP @tab X +@item TCP @tab X +@item UDP @tab X +@end multitable + +@code{X} means that the protocol is supported. + + +@section Input/Output Devices + +@multitable @columnfractions .4 .1 .1 +@item Name @tab Input @tab Output +@item ALSA @tab X @tab X +@item BKTR @tab X @tab +@item DV1394 @tab X @tab +@item JACK @tab X @tab +@item LIBDC1394 @tab X @tab +@item OSS @tab X @tab X +@item Video4Linux @tab X @tab +@item Video4Linux2 @tab X @tab +@item VfW capture @tab X @tab +@item X11 grabbing @tab X @tab +@end multitable + +@code{X} means that input/output is supported. + + +@chapter Platform Specific information + +@section DOS + +Using a cross-compiler is preferred for various reasons. + +@subsection DJGPP + +FFmpeg cannot be compiled because of broken system headers, add +@code{--extra-cflags=-U__STRICT_ANSI__} to the configure options as a +workaround. + +@section OS/2 + +For information about compiling FFmpeg on OS/2 see +@url{http://www.edm2.com/index.php/FFmpeg}. + +@section Unix-like + +Some parts of FFmpeg cannot be built with version 2.15 of the GNU +assembler which is still provided by a few AMD64 distributions. To +make sure your compiler really uses the required version of gas +after a binutils upgrade, run: + +@example +$(gcc -print-prog-name=as) --version +@end example + +If not, then you should install a different compiler that has no +hard-coded path to gas. In the worst case pass @code{--disable-asm} +to configure. + +@subsection BSD + +BSD make will not build FFmpeg, you need to install and use GNU Make +(@file{gmake}). + +@subsubsection FreeBSD + +FreeBSD will not compile out-of-the-box due to broken system headers. +Passing @code{--extra-cflags=-D__BSD_VISIBLE} to configure will work +around the problem. This may have unexpected sideeffects, so use it at +your own risk. If you care about FreeBSD, please make an attempt at +getting the system headers fixed. + +@subsection (Open)Solaris + +GNU Make is required to build FFmpeg, so you have to invoke (@file{gmake}), +standard Solaris Make will not work. When building with a non-c99 front-end +(gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o} +or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options +since the libc is not c99-compliant by default. The probes performed by +configure may raise an exception leading to the death of configure itself +due to a bug in the system shell. Simply invoke a different shell such as +bash directly to work around this: + +@example +bash ./configure +@end example + +@subsection Darwin (MacOS X, iPhone) + +MacOS X on PowerPC or ARM (iPhone) requires a preprocessor from +@url{http://github.com/yuvi/gas-preprocessor} to build the optimized +assembler functions. + +@section Windows + +To get help and instructions for building FFmpeg under Windows, check out +the FFmpeg Windows Help Forum at +@url{http://ffmpeg.arrozcru.org/}. + +@subsection Native Windows compilation + +FFmpeg can be built to run natively on Windows using the MinGW tools. Install +the latest versions of MSYS and MinGW from @url{http://www.mingw.org/}. +You can find detailed installation +instructions in the download section and the FAQ. + +FFmpeg does not build out-of-the-box with the packages the automated MinGW +installer provides. It also requires coreutils to be installed and many other +packages updated to the latest version. The minimum version for some packages +are listed below: + +@itemize +@item bash 3.1 +@item msys-make 3.81-2 (note: not mingw32-make) +@item w32api 3.13 +@item mingw-runtime 3.15 +@end itemize + +FFmpeg automatically passes @code{-fno-common} to the compiler to work around +a GCC bug (see @url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=37216}). + +Within the MSYS shell, configure and make with: + +@example +./configure --enable-memalign-hack +make +make install +@end example + +This will install @file{ffmpeg.exe} along with many other development files +to @file{/usr/local}. You may specify another install path using the +@code{--prefix} option in @file{configure}. + +Notes: + +@itemize + +@item Building natively using MSYS can be sped up by disabling implicit rules +in the Makefile by calling @code{make -r} instead of plain @code{make}. This +speed up is close to non-existent for normal one-off builds and is only +noticeable when running make for a second time (for example in +@code{make install}). + +@item In order to compile FFplay, you must have the MinGW development library +of SDL. Get it from @url{http://www.libsdl.org}. +Edit the @file{bin/sdl-config} script so that it points to the correct prefix +where SDL was installed. Verify that @file{sdl-config} can be launched from +the MSYS command line. + +@item By using @code{./configure --enable-shared} when configuring FFmpeg, +you can build libavutil, libavcodec and libavformat as DLLs. + +@end itemize + +@subsection Microsoft Visual C++ compatibility + +As stated in the FAQ, FFmpeg will not compile under MSVC++. However, if you +want to use the libav* libraries in your own applications, you can still +compile those applications using MSVC++. But the libav* libraries you link +to @emph{must} be built with MinGW. However, you will not be able to debug +inside the libav* libraries, since MSVC++ does not recognize the debug +symbols generated by GCC. +We strongly recommend you to move over from MSVC++ to MinGW tools. + +This description of how to use the FFmpeg libraries with MSVC++ is based on +Microsoft Visual C++ 2005 Express Edition. If you have a different version, +you might have to modify the procedures slightly. + +@subsubsection Using static libraries + +Assuming you have just built and installed FFmpeg in @file{/usr/local}. + +@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 MSVC++ has already created for you. For example, you can copy +@file{libavformat/output-example.c} from the FFmpeg distribution. + +@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 path where the FFmpeg includes were +installed (i.e. @file{c:\msys\1.0\local\include}). +Do not add MinGW's include directory here, or the include files will +conflict with MSVC's. + +@item Still in the "Project / Properties" dialog box, select +"Linker / General" from the tree view and edit the +"Additional Library Directories" setting to contain the @file{lib} +directory where FFmpeg was installed (i.e. @file{c:\msys\1.0\local\lib}), +the directory where MinGW libs are installed (i.e. @file{c:\mingw\lib}), +and the directory where MinGW's GCC libs are installed +(i.e. @file{C:\mingw\lib\gcc\mingw32\4.2.1-sjlj}). Then select +"Linker / Input" from the tree view, and add the files @file{libavformat.a}, +@file{libavcodec.a}, @file{libavutil.a}, @file{libmingwex.a}, +@file{libgcc.a}, and any other libraries you used (i.e. @file{libz.a}) +to the end of "Additional Dependencies". + +@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. + +@item MSVC++ lacks some C99 header files that are fundamental for FFmpeg. +Get msinttypes from @url{http://code.google.com/p/msinttypes/downloads/list} +and install it in MSVC++'s include directory +(i.e. @file{C:\Program Files\Microsoft Visual Studio 8\VC\include}). + +@item MSVC++ also does not understand the @code{inline} keyword used by +FFmpeg, so you must add this line before @code{#include}ing libav*: +@example +#define inline _inline +@end example + +@item Build your application, everything should work. + +@end enumerate + +@subsubsection Using shared libraries + +This is how to create DLL and LIB files that are compatible with MSVC++: + +@enumerate + +@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}: + +@example +call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat" +@end example + +Alternatively, you may start the @file{Visual Studio 2005 Command Prompt}, +and run @file{c:\msys\1.0\msys.bat} from there. + +@item Within the MSYS shell, run @code{lib.exe}. If you get a help message +from @file{Microsoft (R) Library Manager}, this means your environment +variables are set up correctly, the @file{Microsoft (R) Library Manager} +is on the path and will be used by FFmpeg to create +MSVC++-compatible import libraries. + +@item Build FFmpeg with + +@example +./configure --enable-shared --enable-memalign-hack +make +make install +@end example + +Your install path (@file{/usr/local/} by default) should now have the +necessary DLL and LIB files under the @file{bin} directory. + +@end enumerate + +To use those files with MSVC++, do the same as you would do with +the static libraries, as described above. But in Step 4, +you should only need to add the directory where the LIB files are installed +(i.e. @file{c:\msys\usr\local\bin}). This is not a typo, the LIB files are +installed in the @file{bin} directory. And instead of adding @file{libxx.a} +files, you should add @file{avcodec.lib}, @file{avformat.lib}, and +@file{avutil.lib}. There should be no need for @file{libmingwex.a}, +@file{libgcc.a}, and @file{wsock32.lib}, nor any other external library +statically linked into the DLLs. The @file{bin} directory contains a bunch +of DLL files, but the ones that are actually used to run your application +are the ones with a major version number in their filenames +(i.e. @file{avcodec-51.dll}). + +@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 --target-os=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 + +Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack +llrint() in its C library. + +Install your Cygwin with all the "Base" packages, plus the +following "Devel" ones: +@example +binutils, gcc4-core, make, subversion, mingw-runtime, texi2html +@end example + +And the following "Utils" one: +@example +diffutils +@end example + +Then run + +@example +./configure --enable-static --disable-shared +@end example + +to make a static build. + +The current @code{gcc4-core} package is buggy and needs this flag to build +shared libraries: + +@example +./configure --enable-shared --disable-static --extra-cflags=-fno-reorder-functions +@end example + +If you want to build FFmpeg with additional libraries, download Cygwin +"Devel" packages for Ogg and Vorbis from any Cygwin packages repository: +@example +libogg-devel, libvorbis-devel +@end example + +These library packages are only available from Cygwin Ports +(@url{http://sourceware.org/cygwinports/}) : + +@example +yasm, libSDL-devel, libdirac-devel, libfaac-devel, libfaad-devel, libgsm-devel, +libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel, +libxvidcore-devel +@end example + +The recommendation for libnut and x264 is to build them from source by +yourself, as they evolve too quickly for Cygwin Ports to be up to date. + +Cygwin 1.7.x has IPv6 support. You can add IPv6 to Cygwin 1.5.x by means +of the @code{libgetaddrinfo-devel} package, available at Cygwin Ports. + +@subsection Crosscompilation for Windows under Cygwin + +With Cygwin you can create Windows binaries that do not 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 --target-os=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 --target-os=mingw32 --enable-memalign-hack --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin +@end example + +@bye diff --git a/lib/ffmpeg/doc/issue_tracker.txt b/lib/ffmpeg/doc/issue_tracker.txt new file mode 100644 index 0000000000..e5a74db001 --- /dev/null +++ b/lib/ffmpeg/doc/issue_tracker.txt @@ -0,0 +1,228 @@ +FFmpeg's bug/patch/feature request tracker manual +================================================= + +NOTE: This is a draft. + +Overview: +--------- +FFmpeg uses Roundup for tracking issues, new issues and changes to +existing issues can be done through a web interface and through email. +It is possible to subscribe to individual issues by adding yourself to the +nosy list or to subscribe to the ffmpeg-issues mailing list which receives +a mail for every change to every issue. Replies to such mails will also +be properly added to the respective issue. +(the above does all work already after light testing) +The subscription URL for the ffmpeg-issues list is: +http://live.polito/mailman/listinfo/ffmpeg-issues +The URL of the webinterface of the tracker is: +http(s)://roundup.ffmpeg/roundup/ffmpeg/ +Note the URLs in this document are obfuscated, you must append the top level +domain for non-profit organizations to the tracker, and of Italy to the +mailing list. + +Email Interface: +---------------- +There is a mailing list to which all new issues and changes to existing issues +are sent. You can subscribe through +http://live.polito/mailman/listinfo/ffmpeg-issues +Replies to messages there will have their text added to the specific issues. +Attachments will be added as if they had been uploaded via the web interface. +You can change the status, substatus, topic, ... by changing the subject in +your reply like: +Re: [issue94] register_avcodec and allcodecs.h [type=patch;status=open;substatus=approved] +Roundup will then change things as you requested and remove the [...] from +the subject before forwarding the mail to the mailing list. + + +NOTE: issue = (bug report || patch || feature request) + +Type: +----- +bug + An error, flaw, mistake, failure, or fault in FFmpeg or libav* that + prevents it from behaving as intended. + +feature request + Request of support for encoding or decoding of a new codec, container + or variant. + Request of support for more, less or plain different output or behavior + where the current implementation cannot be considered wrong. + +patch + A patch as generated by diff which conforms to the patch submission and + development policy. + + +Priority: +--------- +critical + Bugs and patches which deal with data loss and security issues. + No feature request can be critical. + +important + Bugs which make FFmpeg unusable for a significant number of users, and + patches fixing them. + Examples here might be completely broken MPEG-4 decoding or a build issue + on Linux. + While broken 4xm decoding or a broken OS/2 build would not be important, + the separation to normal is somewhat fuzzy. + For feature requests this priority would be used for things many people + want. + +normal + + +minor + Bugs and patches about things like spelling errors, "mp2" instead of + "mp3" being shown and such. + Feature requests about things few people want or which do not make a big + difference. + +wish + Something that is desirable to have but that there is no urgency at + all to implement, e.g. something completely cosmetic like a website + restyle or a personalized doxy template or the FFmpeg logo. + This priority is not valid for bugs. + + +Status: +------- +new + initial state + +open + intermediate states + +closed + final state + + +Type/Status/Substatus: +---------- +*/new/new + Initial state of new bugs, patches and feature requests submitted by + users. + +*/open/open + Issues which have been briefly looked at and which did not look outright + invalid. + This implicates that no real more detailed state applies yet. Conversely, + the more detailed states below implicate that the issue has been briefly + looked at. + +*/closed/duplicate + Bugs, patches or feature requests which are duplicates. + Note that patches dealing with the same thing in a different way are not + duplicates. + Note, if you mark something as duplicate, do not forget setting the + superseder so bug reports are properly linked. + +*/closed/invalid + Bugs caused by user errors, random ineligible or otherwise nonsense stuff. + +*/closed/needs_more_info + Issues for which some information has been requested by the developers, + but which has not been provided by anyone within reasonable time. + +bug/open/reproduced + Bugs which have been reproduced. + +bug/open/analyzed + Bugs which have been analyzed and where it is understood what causes them + and which exact chain of events triggers them. This analysis should be + available as a message in the bug report. + Note, do not change the status to analyzed without also providing a clear + and understandable analysis. + This state implicates that the bug either has been reproduced or that + reproduction is not needed as the bug is already understood. + +bug/open/needs_more_info + Bug reports which are incomplete and or where more information is needed + from the submitter or another person who can provide it. + This state implicates that the bug has not been analyzed or reproduced. + Note, the idea behind needs_more_info is to offload work from the + developers to the users whenever possible. + +bug/closed/fixed + Bugs which have to the best of our knowledge been fixed. + +bug/closed/wont_fix + Bugs which we will not fix. Possible reasons include legality, high + complexity for the sake of supporting obscure corner cases, speed loss + for similarly esoteric purposes, et cetera. + This also means that we would reject a patch. + If we are just too lazy to fix a bug then the correct state is open + and unassigned. Closed means that the case is closed which is not + the case if we are just waiting for a patch. + +bug/closed/works_for_me + Bugs for which sufficient information was provided to reproduce but + reproduction failed - that is the code seems to work correctly to the + best of our knowledge. + +patch/open/approved + Patches which have been reviewed and approved by a developer. + Such patches can be applied anytime by any other developer after some + reasonable testing (compile + regression tests + does the patch do + what the author claimed). + +patch/open/needs_changes + Patches which have been reviewed and need changes to be accepted. + +patch/closed/applied + Patches which have been applied. + +patch/closed/rejected + Patches which have been rejected. + +feature_request/open/needs_more_info + Feature requests where it is not clear what exactly is wanted + (these also could be closed as invalid ...). + +feature_request/closed/implemented + Feature requests which have been implemented. + +feature_request/closed/wont_implement + Feature requests which will not be implemented. The reasons here could + be legal, philosophical or others. + +Note, please do not use type-status-substatus combinations other than the +above without asking on ffmpeg-dev first! + +Note2, if you provide the requested info do not forget to remove the +needs_more_info substate. + +Topic: +------ +A topic is a tag you should add to your issue in order to make grouping them +easier. + +avcodec + issues in libavcodec/* + +avformat + issues in libavformat/* + +avutil + issues in libavutil/* + +regression test + issues in tests/* + +ffmpeg + issues in or related to ffmpeg.c + +ffplay + issues in or related to ffplay.c + +ffserver + issues in or related to ffserver.c + +build system + issues in or related to configure/Makefile + +regression + bugs which were working in a past revision + +roundup + issues related to our issue tracker diff --git a/lib/ffmpeg/doc/libavfilter.texi b/lib/ffmpeg/doc/libavfilter.texi new file mode 100644 index 0000000000..8745928d40 --- /dev/null +++ b/lib/ffmpeg/doc/libavfilter.texi @@ -0,0 +1,104 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libavfilter Documentation +@titlepage +@sp 7 +@center @titlefont{Libavfilter Documentation} +@sp 3 +@end titlepage + + +@chapter Introduction + +Libavfilter is the filtering API of FFmpeg. It is the substitute of the +now deprecated 'vhooks' and started as a Google Summer of Code project. + +Integrating libavfilter into the main FFmpeg repository is a work in +progress. If you wish to try the unfinished development code of +libavfilter then check it out from the libavfilter repository into +some directory of your choice by: + +@example + svn checkout svn://svn.ffmpeg.org/soc/libavfilter +@end example + +And then read the README file in the top directory to learn how to +integrate it into ffmpeg and ffplay. + +But note that there may still be serious bugs in the code and its API +and ABI should not be considered stable yet! + +@chapter Tutorial + +In libavfilter, it is possible for filters to have multiple inputs and +multiple outputs. +To illustrate the sorts of things that are possible, we can +use a complex filter graph. For example, the following one: + +@example +input --> split --> fifo -----------------------> overlay --> output + | ^ + | | + +------> fifo --> crop --> vflip --------+ +@end example + +splits the stream in two streams, sends one stream through the crop filter +and the vflip filter before merging it back with the other stream by +overlaying it on top. You can use the following command to achieve this: + +@example +./ffmpeg -i in.avi -s 240x320 -vf "[in] split [T1], fifo, [T2] overlay= 0:240 [out]; [T1] fifo, crop=0:0:-1:240, vflip [T2] +@end example + +where input_video.avi has a vertical resolution of 480 pixels. The +result will be that in output the top half of the video is mirrored +onto the bottom half. + +Video filters are loaded using the @var{-vf} option passed to +ffmpeg or to ffplay. Filters in the same linear chain are separated by +commas. In our example, @var{split, fifo, overlay} are in one linear +chain, and @var{fifo, crop, vflip} are in another. The points where +the linear chains join are labeled by names enclosed in square +brackets. In our example, that is @var{[T1]} and @var{[T2]}. The magic +labels @var{[in]} and @var{[out]} are the points where video is input +and output. + +Some filters take in input a list of parameters: they are specified +after the filter name and an equal sign, and are separated each other +by a semicolon. + +There exist so-called @var{source filters} that do not have a video +input, and we expect in the future some @var{sink filters} that will +not have video output. + +@chapter graph2dot + +The @file{graph2dot} program included in the FFmpeg @file{tools} +directory can be used to parse a filter graph description and issue a +corresponding textual representation in the dot language. + +Invoke the command: +@example +graph2dot -h +@end example + +to see how to use @file{graph2dot}. + +You can then pass the dot description to the @file{dot} program (from +the graphviz suite of programs) and obtain a graphical representation +of the filter graph. + +For example the sequence of commands: +@example +echo @var{GRAPH_DESCRIPTION} | \ +tools/graph2dot -o graph.tmp && \ +dot -Tpng graph.tmp -o graph.png && \ +display graph.png +@end example + +can be used to create and display an image representing the graph +described by the @var{GRAPH_DESCRIPTION} string. + +@include filters.texi + +@bye diff --git a/lib/ffmpeg/doc/optimization.txt b/lib/ffmpeg/doc/optimization.txt new file mode 100644 index 0000000000..5469adc836 --- /dev/null +++ b/lib/ffmpeg/doc/optimization.txt @@ -0,0 +1,235 @@ +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 x86/ 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 x86 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 recommended to look at older +revisions of the interesting files (for a web frontend try ViewVC at +http://svn.ffmpeg.org/ffmpeg/trunk/). +Alternatively, look into the other architecture-specific versions in +the x86/, 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) + + +When is an optimization justified? +---------------------------------- +Normally, clean and simple optimizations for widely used codecs are +justified even if they only achieve an overall speedup of 0.1%. These +speedups accumulate and can make a big difference after awhile. Also, if +none of the following factors get worse due to an optimization -- speed, +binary code size, source size, source readability -- and at least one +factor improves, then an optimization is always a good idea even if the +overall gain is less than 0.1%. For obscure codecs that are not often +used, the goal is more toward keeping the code clean, small, and +readable instead of making it 1% faster. + + +WTF is that function good for ....: +----------------------------------- +The primary purpose of this list is to avoid wasting time optimizing 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. + +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); + + +General Tips: +------------- +Use asm loops like: +__asm__( + "1: .... + ... + "jump_instruciton .... +Do not use C loops: +do{ + __asm__( + ... +}while() + +Use __asm__() instead of intrinsics. The latter requires a good optimizing compiler +which gcc is not. + + +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 + + +ARM-specific: +------------- +ARM Architecture Reference Manual (up to ARMv5TE): +http://www.arm.com/community/university/eulaarmarm.html + +Procedure Call Standard for the ARM Architecture: +http://www.arm.com/pdfs/aapcs.pdf + +Optimization guide for ARM9E (used in Nokia 770 Internet Tablet): +http://infocenter.arm.com/help/topic/com.arm.doc.ddi0240b/DDI0240A.pdf +Optimization guide for ARM11 (used in Nokia N800 Internet Tablet): +http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf +Optimization guide for Intel XScale (used in Sharp Zaurus PDA): +http://download.intel.com/design/intelxscale/27347302.pdf +Intel Wireless MMX2 Coprocessor: Programmers Reference Manual +http://download.intel.com/design/intelxscale/31451001.pdf + +PowerPC-specific: +----------------- +PowerPC32/AltiVec PIM: +www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPEM.pdf + +PowerPC32/AltiVec PEM: +www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPIM.pdf + +CELL/SPU: +http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf +http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf + +SPARC-specific: +--------------- +SPARC Joint Programming Specification (JPS1): Commonality +http://www.fujitsu.com/downloads/PRMPWR/JPS1-R1.0.4-Common-pub.pdf + +UltraSPARC III Processor User's Manual (contains instruction timings) +http://www.sun.com/processors/manuals/USIIIv2.pdf + +VIS Whitepaper (contains optimization guidelines) +http://www.sun.com/processors/vis/download/vis/vis_whitepaper.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/lib/ffmpeg/doc/rate_distortion.txt b/lib/ffmpeg/doc/rate_distortion.txt new file mode 100644 index 0000000000..a7d2c878b2 --- /dev/null +++ b/lib/ffmpeg/doc/rate_distortion.txt @@ -0,0 +1,61 @@ +A Quick Description Of Rate Distortion Theory. + +We want to encode a video, picture or piece of music optimally. What does +"optimally" really mean? It means that we want to get the best quality at a +given filesize OR we want to get the smallest filesize at a given quality +(in practice, these 2 goals are usually the same). + +Solving this directly is not practical; trying all byte sequences 1 +megabyte in length and selecting the "best looking" sequence will yield +256^1000000 cases to try. + +But first, a word about quality, which is also called distortion. +Distortion can be quantified by almost any quality measurement one chooses. +Commonly, the sum of squared differences is used but more complex methods +that consider psychovisual effects can be used as well. It makes no +difference in this discussion. + + +First step: that rate distortion factor called lambda... +Let's consider the problem of minimizing: + + distortion + lambda*rate + +rate is the filesize +distortion is the quality +lambda is a fixed value choosen as a tradeoff between quality and filesize +Is this equivalent to finding the best quality for a given max +filesize? The answer is yes. For each filesize limit there is some lambda +factor for which minimizing above will get you the best quality (using your +chosen quality measurement) at the desired (or lower) filesize. + + +Second step: splitting the problem. +Directly splitting the problem of finding the best quality at a given +filesize is hard because we do not know how many bits from the total +filesize should be allocated to each of the subproblems. But the formula +from above: + + distortion + lambda*rate + +can be trivially split. Consider: + + (distortion0 + distortion1) + lambda*(rate0 + rate1) + +This creates a problem made of 2 independent subproblems. The subproblems +might be 2 16x16 macroblocks in a frame of 32x16 size. To minimize: + + (distortion0 + distortion1) + lambda*(rate0 + rate1) + +we just have to minimize: + + distortion0 + lambda*rate0 + +and + + distortion1 + lambda*rate1 + +I.e, the 2 problems can be solved independently. + +Author: Michael Niedermayer +Copyright: LGPL diff --git a/lib/ffmpeg/doc/snow.txt b/lib/ffmpeg/doc/snow.txt new file mode 100644 index 0000000000..f99133971c --- /dev/null +++ b/lib/ffmpeg/doc/snow.txt @@ -0,0 +1,630 @@ +============================================= +Snow Video Codec Specification Draft 20080110 +============================================= + +Introduction: +============= +This specification describes the Snow bitstream syntax and semantics as +well as the formal Snow decoding process. + +The decoding process is described precisely and any compliant decoder +MUST produce the exact same output for a spec-conformant Snow stream. +For encoding, though, any process which generates a stream compliant to +the syntactical and semantic requirements and which is decodable by +the process described in this spec shall be considered a conformant +Snow encoder. + +Definitions: +============ + +MUST the specific part must be done to conform to this standard +SHOULD it is recommended to be done that way, but not strictly required + +ilog2(x) is the rounded down logarithm of x with basis 2 +ilog2(0) = 0 + +Type definitions: +================= + +b 1-bit range coded +u unsigned scalar value range coded +s signed scalar value range coded + + +Bitstream syntax: +================= + +frame: + header + prediction + residual + +header: + keyframe b MID_STATE + if(keyframe || always_reset) + reset_contexts + if(keyframe){ + version u header_state + always_reset b header_state + temporal_decomposition_type u header_state + temporal_decomposition_count u header_state + spatial_decomposition_count u header_state + colorspace_type u header_state + chroma_h_shift u header_state + chroma_v_shift u header_state + spatial_scalability b header_state + max_ref_frames-1 u header_state + qlogs + } + if(!keyframe){ + update_mc b header_state + if(update_mc){ + for(plane=0; plane<2; plane++){ + diag_mc b header_state + htaps/2-1 u header_state + for(i= p->htaps/2; i; i--) + |hcoeff[i]| u header_state + } + } + update_qlogs b header_state + if(update_qlogs){ + spatial_decomposition_count u header_state + qlogs + } + } + + spatial_decomposition_type s header_state + qlog s header_state + mv_scale s header_state + qbias s header_state + block_max_depth s header_state + +qlogs: + for(plane=0; plane<2; plane++){ + quant_table[plane][0][0] s header_state + for(level=0; level < spatial_decomposition_count; level++){ + quant_table[plane][level][1]s header_state + quant_table[plane][level][3]s header_state + } + } + +reset_contexts + *_state[*]= MID_STATE + +prediction: + for(y=0; y<block_count_vertical; y++) + for(x=0; x<block_count_horizontal; x++) + block(0) + +block(level): + mvx_diff=mvy_diff=y_diff=cb_diff=cr_diff=0 + if(keyframe){ + intra=1 + }else{ + if(level!=max_block_depth){ + s_context= 2*left->level + 2*top->level + topleft->level + topright->level + leaf b block_state[4 + s_context] + } + if(level==max_block_depth || leaf){ + intra b block_state[1 + left->intra + top->intra] + if(intra){ + y_diff s block_state[32] + cb_diff s block_state[64] + cr_diff s block_state[96] + }else{ + ref_context= ilog2(2*left->ref) + ilog2(2*top->ref) + if(ref_frames > 1) + ref u block_state[128 + 1024 + 32*ref_context] + mx_context= ilog2(2*abs(left->mx - top->mx)) + my_context= ilog2(2*abs(left->my - top->my)) + mvx_diff s block_state[128 + 32*(mx_context + 16*!!ref)] + mvy_diff s block_state[128 + 32*(my_context + 16*!!ref)] + } + }else{ + block(level+1) + block(level+1) + block(level+1) + block(level+1) + } + } + + +residual: + residual2(luma) + residual2(chroma_cr) + residual2(chroma_cb) + +residual2: + for(level=0; level<spatial_decomposition_count; level++){ + if(level==0) + subband(LL, 0) + subband(HL, level) + subband(LH, level) + subband(HH, level) + } + +subband: + FIXME + + + +Tag description: +---------------- + +version + 0 + this MUST NOT change within a bitstream + +always_reset + if 1 then the range coder contexts will be reset after each frame + +temporal_decomposition_type + 0 + +temporal_decomposition_count + 0 + +spatial_decomposition_count + FIXME + +colorspace_type + 0 + this MUST NOT change within a bitstream + +chroma_h_shift + log2(luma.width / chroma.width) + this MUST NOT change within a bitstream + +chroma_v_shift + log2(luma.height / chroma.height) + this MUST NOT change within a bitstream + +spatial_scalability + 0 + +max_ref_frames + maximum number of reference frames + this MUST NOT change within a bitstream + +update_mc + indicates that motion compensation filter parameters are stored in the + header + +diag_mc + flag to enable faster diagonal interpolation + this SHOULD be 1 unless it turns out to be covered by a valid patent + +htaps + number of half pel interpolation filter taps, MUST be even, >0 and <10 + +hcoeff + half pel interpolation filter coefficients, hcoeff[0] are the 2 middle + coefficients [1] are the next outer ones and so on, resulting in a filter + like: ...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... + the sign of the coefficients is not explicitly stored but alternates + after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... + hcoeff[0] is not explicitly stored but found by subtracting the sum + of all stored coefficients with signs from 32 + hcoeff[0]= 32 - hcoeff[1] - hcoeff[2] - ... + a good choice for hcoeff and htaps is + htaps= 6 + hcoeff={40,-10,2} + an alternative which requires more computations at both encoder and + decoder side and may or may not be better is + htaps= 8 + hcoeff={42,-14,6,-2} + + +ref_frames + minimum of the number of available reference frames and max_ref_frames + for example the first frame after a key frame always has ref_frames=1 + +spatial_decomposition_type + wavelet type + 0 is a 9/7 symmetric compact integer wavelet + 1 is a 5/3 symmetric compact integer wavelet + others are reserved + stored as delta from last, last is reset to 0 if always_reset || keyframe + +qlog + quality (logarthmic quantizer scale) + stored as delta from last, last is reset to 0 if always_reset || keyframe + +mv_scale + stored as delta from last, last is reset to 0 if always_reset || keyframe + FIXME check that everything works fine if this changes between frames + +qbias + dequantization bias + stored as delta from last, last is reset to 0 if always_reset || keyframe + +block_max_depth + maximum depth of the block tree + stored as delta from last, last is reset to 0 if always_reset || keyframe + +quant_table + quantiztation table + + +Highlevel bitstream structure: +============================= + -------------------------------------------- +| Header | + -------------------------------------------- +| ------------------------------------ | +| | Block0 | | +| | split? | | +| | yes no | | +| | ......... intra? | | +| | : Block01 : yes no | | +| | : Block02 : ....... .......... | | +| | : Block03 : : y DC : : ref index: | | +| | : Block04 : : cb DC : : motion x : | | +| | ......... : cr DC : : motion y : | | +| | ....... .......... | | +| ------------------------------------ | +| ------------------------------------ | +| | Block1 | | +| ... | + -------------------------------------------- +| ------------ ------------ ------------ | +|| Y subbands | | Cb subbands| | Cr subbands|| +|| --- --- | | --- --- | | --- --- || +|| |LL0||HL0| | | |LL0||HL0| | | |LL0||HL0| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |LH0||HH0| | | |LH0||HH0| | | |LH0||HH0| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |HL1||LH1| | | |HL1||LH1| | | |HL1||LH1| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |HH1||HL2| | | |HH1||HL2| | | |HH1||HL2| || +|| ... | | ... | | ... || +| ------------ ------------ ------------ | + -------------------------------------------- + +Decoding process: +================= + + ------------ + | | + | Subbands | + ------------ | | + | | ------------ + | Intra DC | | + | | LL0 subband prediction + ------------ | + \ Dequantizaton + ------------------- \ | +| Reference frames | \ IDWT +| ------- ------- | Motion \ | +||Frame 0| |Frame 1|| Compensation . OBMC v ------- +| ------- ------- | --------------. \------> + --->|Frame n|-->output +| ------- ------- | ------- +||Frame 2| |Frame 3||<----------------------------------/ +| ... | + ------------------- + + +Range Coder: +============ + +Binary Range Coder: +------------------- +The implemented range coder is an adapted version based upon "Range encoding: +an algorithm for removing redundancy from a digitised message." by G. N. N. +Martin. +The symbols encoded by the Snow range coder are bits (0|1). The +associated probabilities are not fix but change depending on the symbol mix +seen so far. + + +bit seen | new state +---------+----------------------------------------------- + 0 | 256 - state_transition_table[256 - old_state]; + 1 | state_transition_table[ old_state]; + +state_transition_table = { + 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, + 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, + 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, + 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, + 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, + 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, +104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, +119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, +134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, +150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, +165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, +180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, +195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, +210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, +226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, +241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0}; + +FIXME + + +Range Coding of integers: +------------------------- +FIXME + + +Neighboring Blocks: +=================== +left and top are set to the respective blocks unless they are outside of +the image in which case they are set to the Null block + +top-left is set to the top left block unless it is outside of the image in +which case it is set to the left block + +if this block has no larger parent block or it is at the left side of its +parent block and the top right block is not outside of the image then the +top right block is used for top-right else the top-left block is used + +Null block +y,cb,cr are 128 +level, ref, mx and my are 0 + + +Motion Vector Prediction: +========================= +1. the motion vectors of all the neighboring blocks are scaled to +compensate for the difference of reference frames + +scaled_mv= (mv * (256 * (current_reference+1) / (mv.reference+1)) + 128)>>8 + +2. the median of the scaled left, top and top-right vectors is used as +motion vector prediction + +3. the used motion vector is the sum of the predictor and + (mvx_diff, mvy_diff)*mv_scale + + +Intra DC Predicton: +====================== +the luma and chroma values of the left block are used as predictors + +the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff +to reverse this in the decoder apply the following: +block[y][x].dc[0] = block[y][x-1].dc[0] + y_diff; +block[y][x].dc[1] = block[y][x-1].dc[1] + cb_diff; +block[y][x].dc[2] = block[y][x-1].dc[2] + cr_diff; +block[*][-1].dc[*]= 128; + + +Motion Compensation: +==================== + +Halfpel interpolation: +---------------------- +halfpel interpolation is done by convolution with the halfpel filter stored +in the header: + +horizontal halfpel samples are found by +H1[y][x] = hcoeff[0]*(F[y][x ] + F[y][x+1]) + + hcoeff[1]*(F[y][x-1] + F[y][x+2]) + + hcoeff[2]*(F[y][x-2] + F[y][x+3]) + + ... +h1[y][x] = (H1[y][x] + 32)>>6; + +vertical halfpel samples are found by +H2[y][x] = hcoeff[0]*(F[y ][x] + F[y+1][x]) + + hcoeff[1]*(F[y-1][x] + F[y+2][x]) + + ... +h2[y][x] = (H2[y][x] + 32)>>6; + +vertical+horizontal halfpel samples are found by +H3[y][x] = hcoeff[0]*(H2[y][x ] + H2[y][x+1]) + + hcoeff[1]*(H2[y][x-1] + H2[y][x+2]) + + ... +H3[y][x] = hcoeff[0]*(H1[y ][x] + H1[y+1][x]) + + hcoeff[1]*(H1[y+1][x] + H1[y+2][x]) + + ... +h3[y][x] = (H3[y][x] + 2048)>>12; + + + F H1 F + | | | + | | | + | | | + F H1 F + | | | + | | | + | | | + F-------F-------F-> H1<-F-------F-------F + v v v + H2 H3 H2 + ^ ^ ^ + F-------F-------F-> H1<-F-------F-------F + | | | + | | | + | | | + F H1 F + | | | + | | | + | | | + F H1 F + + +unavailable fullpel samples (outside the picture for example) shall be equal +to the closest available fullpel sample + + +Smaller pel interpolation: +-------------------------- +if diag_mc is set then points which lie on a line between 2 vertically, +horiziontally or diagonally adjacent halfpel points shall be interpolated +linearls with rounding to nearest and halfway values rounded up. +points which lie on 2 diagonals at the same time should only use the one +diagonal not containing the fullpel point + + + + F-->O---q---O<--h1->O---q---O<--F + v \ / v \ / v + O O O O O O O + | / | \ | + q q q q q + | / | \ | + O O O O O O O + ^ / \ ^ / \ ^ + h2-->O---q---O<--h3->O---q---O<--h2 + v \ / v \ / v + O O O O O O O + | \ | / | + q q q q q + | \ | / | + O O O O O O O + ^ / \ ^ / \ ^ + F-->O---q---O<--h1->O---q---O<--F + + + +the remaining points shall be bilinearly interpolated from the +up to 4 surrounding halfpel and fullpel points, again rounding should be to +nearest and halfway values rounded up + +compliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chroma +interpolation at least + + +Overlapped block motion compensation: +------------------------------------- +FIXME + +LL band prediction: +=================== +Each sample in the LL0 subband is predicted by the median of the left, top and +left+top-topleft samples, samples outside the subband shall be considered to +be 0. To reverse this prediction in the decoder apply the following. +for(y=0; y<height; y++){ + for(x=0; x<width; x++){ + sample[y][x] += median(sample[y-1][x], + sample[y][x-1], + sample[y-1][x]+sample[y][x-1]-sample[y-1][x-1]); + } +} +sample[-1][*]=sample[*][-1]= 0; +width,height here are the width and height of the LL0 subband not of the final +video + + +Dequantizaton: +============== +FIXME + +Wavelet Transform: +================== + +Snow supports 2 wavelet transforms, the symmetric biorthogonal 5/3 integer +transform and a integer approximation of the symmetric biorthogonal 9/7 +daubechies wavelet. + +2D IDWT (inverse discrete wavelet transform) +-------------------------------------------- +The 2D IDWT applies a 2D filter recursively, each time combining the +4 lowest frequency subbands into a single subband until only 1 subband +remains. +The 2D filter is done by first applying a 1D filter in the vertical direction +and then applying it in the horizontal one. + --------------- --------------- --------------- --------------- +|LL0|HL0| | | | | | | | | | | | +|---+---| HL1 | | L0|H0 | HL1 | | LL1 | HL1 | | | | +|LH0|HH0| | | | | | | | | | | | +|-------+-------|->|-------+-------|->|-------+-------|->| L1 | H1 |->... +| | | | | | | | | | | | +| LH1 | HH1 | | LH1 | HH1 | | LH1 | HH1 | | | | +| | | | | | | | | | | | + --------------- --------------- --------------- --------------- + + +1D Filter: +---------- +1. interleave the samples of the low and high frequency subbands like +s={L0, H0, L1, H1, L2, H2, L3, H3, ... } +note, this can end with a L or a H, the number of elements shall be w +s[-1] shall be considered equivalent to s[1 ] +s[w ] shall be considered equivalent to s[w-2] + +2. perform the lifting steps in order as described below + +5/3 Integer filter: +1. s[i] -= (s[i-1] + s[i+1] + 2)>>2; for all even i < w +2. s[i] += (s[i-1] + s[i+1] )>>1; for all odd i < w + +\ | /|\ | /|\ | /|\ | /|\ + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | -1/4 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + | + | + | + | + +1/2 + + +Snow's 9/7 Integer filter: +1. s[i] -= (3*(s[i-1] + s[i+1]) + 4)>>3; for all even i < w +2. s[i] -= s[i-1] + s[i+1] ; for all odd i < w +3. s[i] += ( s[i-1] + s[i+1] + 4*s[i] + 8)>>4; for all even i < w +4. s[i] += (3*(s[i-1] + s[i+1]) )>>1; for all odd i < w + +\ | /|\ | /|\ | /|\ | /|\ + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | -3/8 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + (| + (| + (| + (| + -1 +\ + /|\ + /|\ + /|\ + /|\ +1/4 + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | +1/16 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + | + | + | + | + +3/2 + +optimization tips: +following are exactly identical +(3a)>>1 == a + (a>>1) +(a + 4b + 8)>>4 == ((a>>2) + b + 2)>>2 + +16bit implementation note: +The IDWT can be implemented with 16bits, but this requires some care to +prevent overflows, the following list, lists the minimum number of bits needed +for some terms +1. lifting step +A= s[i-1] + s[i+1] 16bit +3*A + 4 18bit +A + (A>>1) + 2 17bit + +3. lifting step +s[i-1] + s[i+1] 17bit + +4. lifiting step +3*(s[i-1] + s[i+1]) 17bit + + +TODO: +===== +Important: +finetune initial contexts +flip wavelet? +try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients +try the MV length as context for coding the residual coefficients +use extradata for stuff which is in the keyframes now? +the MV median predictor is patented IIRC +implement per picture halfpel interpolation +try different range coder state transition tables for different contexts + +Not Important: +compare the 6 tap and 8 tap hpel filters (psnr/bitrate and subjective quality) +spatial_scalability b vs u (!= 0 breaks syntax anyway so we can add a u later) + + +Credits: +======== +Michael Niedermayer +Loren Merritt + + +Copyright: +========== +GPL + GFDL + whatever is needed to make this a RFC diff --git a/lib/ffmpeg/doc/soc.txt b/lib/ffmpeg/doc/soc.txt new file mode 100644 index 0000000000..8b4a86db80 --- /dev/null +++ b/lib/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/lib/ffmpeg/doc/swscale.txt b/lib/ffmpeg/doc/swscale.txt new file mode 100644 index 0000000000..4c62e67321 --- /dev/null +++ b/lib/ffmpeg/doc/swscale.txt @@ -0,0 +1,99 @@ + The official guide to swscale for confused developers. + ======================================================== + +Current (simplified) Architecture: +--------------------------------- + Input + v + _______OR_________ + / \ + / \ + special converter [Input to YUV converter] + | | + | (8bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:0:0 ) + | | + | v + | Horizontal scaler + | | + | (15bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:1:1 / 4:0:0 ) + | | + | v + | Vertical scaler and output converter + | | + v v + output + + +Swscale has 2 scaler paths. Each side must be capable of handling +slices, that is, consecutive non-overlapping rectangles of dimension +(0,slice_top) - (picture_width, slice_bottom). + +special converter + These generally are unscaled converters of common + formats, like YUV 4:2:0/4:2:2 -> RGB12/15/16/24/32. Though it could also + in principle contain scalers optimized for specific common cases. + +Main path + The main path is used when no special converter can be used. The code + is designed as a destination line pull architecture. That is, for each + output line the vertical scaler pulls lines from a ring buffer. When + the ring buffer does not contain the wanted line, then it is pulled from + the input slice through the input converter and horizontal scaler. + The result is also stored in the ring buffer to serve future vertical + scaler requests. + When no more output can be generated because lines from a future slice + would be needed, then all remaining lines in the current slice are + converted, horizontally scaled and put in the ring buffer. + [This is done for luma and chroma, each with possibly different numbers + of lines per picture.] + +Input to YUV Converter + When the input to the main path is not planar 8 bits per component YUV or + 8-bit gray, it is converted to planar 8-bit YUV. Two sets of converters + exist for this currently: One performs horizontal downscaling by 2 + before the conversion, the other leaves the full chroma resolution, + but is slightly slower. The scaler will try to preserve full chroma + when the output uses it. It is possible to force full chroma with + SWS_FULL_CHR_H_INP even for cases where the scaler thinks it is useless. + +Horizontal scaler + There are several horizontal scalers. A special case worth mentioning is + the fast bilinear scaler that is made of runtime-generated MMX2 code + using specially tuned pshufw instructions. + The remaining scalers are specially-tuned for various filter lengths. + They scale 8-bit unsigned planar data to 16-bit signed planar data. + Future >8 bits per component inputs will need to add a new horizontal + scaler that preserves the input precision. + +Vertical scaler and output converter + There is a large number of combined vertical scalers + output converters. + Some are: + * unscaled output converters + * unscaled output converters that average 2 chroma lines + * bilinear converters (C, MMX and accurate MMX) + * arbitrary filter length converters (C, MMX and accurate MMX) + And + * Plain C 8-bit 4:2:2 YUV -> RGB converters using LUTs + * Plain C 17-bit 4:4:4 YUV -> RGB converters using multiplies + * MMX 11-bit 4:2:2 YUV -> RGB converters + * Plain C 16-bit Y -> 16-bit gray + ... + + RGB with less than 8 bits per component uses dither to improve the + subjective quality and low-frequency accuracy. + + +Filter coefficients: +-------------------- +There are several different scalers (bilinear, bicubic, lanczos, area, +sinc, ...). Their coefficients are calculated in initFilter(). +Horizontal filter coefficients have a 1.0 point at 1 << 14, vertical ones at +1 << 12. The 1.0 points have been chosen to maximize precision while leaving +a little headroom for convolutional filters like sharpening filters and +minimizing SIMD instructions needed to apply them. +It would be trivial to use a different 1.0 point if some specific scaler +would benefit from it. +Also, as already hinted at, initFilter() accepts an optional convolutional +filter as input that can be used for contrast, saturation, blur, sharpening +shift, chroma vs. luma shift, ... + diff --git a/lib/ffmpeg/doc/tablegen.txt b/lib/ffmpeg/doc/tablegen.txt new file mode 100644 index 0000000000..4c4f036e6a --- /dev/null +++ b/lib/ffmpeg/doc/tablegen.txt @@ -0,0 +1,70 @@ +Writing a table generator + +This documentation is preliminary. +Parts of the API are not good and should be changed. + +Basic concepts + +A table generator consists of two files, *_tablegen.c and *_tablegen.h. +The .h file will provide the variable declarations and initialization +code for the tables, the .c calls the initialization code and then prints +the tables as a header file using the tableprint.h helpers. +Both of these files will be compiled for the host system, so to avoid +breakage with cross-compilation neither of them may include, directly +or indirectly, config.h or avconfig.h. +This means that e.g. libavutil/mathematics.h is ok but libavutil/libm.h is not. +Due to this, the .c file or Makefile may have to provide additional defines +or stubs, though if possible this should be avoided. +In particular, CONFIG_HARDCODED_TABLES should always be defined to 0. + +The .c file + +This file should include the *_tablegen.h and tableprint.h files and +anything else it needs as long as it does not depend on config.h or +avconfig.h. +In addition to that it must contain a main() function which initializes +all tables by calling the init functions from the .h file and then prints +them. +The printing code typically looks like this: + write_fileheader(); + printf("static const uint8_t my_array[100] = {\n"); + write_uint8_t_array(my_array, 100); + printf("};\n"); + +This is the more generic form, in case you need to do something special. +Usually you should instead use the short form: + write_fileheader(); + WRITE_ARRAY("static const", uint8_t, my_array); + +write_fileheader() adds some minor things like a "this is a generated file" +comment and some standard includes. +tablegen.h defines some write functions for one- and two-dimensional arrays +for standard types - they print only the "core" parts so they are easier +to reuse for multi-dimensional arrays so the outermost {} must be printed +separately. +If there's no standard function for printing the type you need, the +WRITE_1D_FUNC_ARGV macro is a very quick way to create one. +See libavcodec/dv_tablegen.c for an example. + + +The .h file + +This file should contain: + - one or more initialization functions + - the table variable declarations +If CONFIG_HARDCODED_TABLES is set, the initialization functions should +not do anything, and instead of the variable declarations the +generated *_tables.h file should be included. +Since that will be generated in the build directory, the path must be +included, i.e. +#include "libavcodec/example_tables.h" +not +#include "example_tables.h" + +Makefile changes + +To make the automatic table creation work, you must manually declare the +new dependency. +For this add a line similar to this: +$(SUBDIR)example.o: $(SUBDIR)example_tables.h +under the "ifdef CONFIG_HARDCODED_TABLES" section in the Makefile. diff --git a/lib/ffmpeg/doc/texi2pod.pl b/lib/ffmpeg/doc/texi2pod.pl new file mode 100755 index 0000000000..fd3f02059d --- /dev/null +++ b/lib/ffmpeg/doc/texi2pod.pl @@ -0,0 +1,423 @@ +#! /usr/bin/perl -w + +# Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc. + +# This file is part of GNU CC. + +# GNU CC is free software; you can redistribute it and/or modify +# it under the terms of the GNU General Public License as published by +# the Free Software Foundation; either version 2, or (at your option) +# any later version. + +# GNU CC is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# GNU General Public License for more details. + +# You should have received a copy of the GNU General Public License +# along with GNU CC; see the file COPYING. If not, write to +# the Free Software Foundation, 51 Franklin Street, Fifth Floor, +# Boston, MA 02110-1301 USA + +# This does trivial (and I mean _trivial_) conversion of Texinfo +# markup to Perl POD format. It's intended to be used to extract +# something suitable for a manpage from a Texinfo document. + +$output = 0; +$skipping = 0; +%sects = (); +@sects_sequence = (); +$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($2); + } + next; + }; + + /^\@include\s+(.+)$/ 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; + }; + + # 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-Za-z ]+)/ and $sect = $1, push (@sects_sequence, $sect), $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. + + /^\@(?: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 ? $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}; + +unshift @sects_sequence, "NAME"; +for $sect (@sects_sequence) { + 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; + } +} diff --git a/lib/ffmpeg/doc/viterbi.txt b/lib/ffmpeg/doc/viterbi.txt new file mode 100644 index 0000000000..d9d924f621 --- /dev/null +++ b/lib/ffmpeg/doc/viterbi.txt @@ -0,0 +1,110 @@ +This is a quick description of the viterbi aka dynamic programing +algorthm. + +Its reason for existence is that wikipedia has become very poor on +describing algorithms in a way that makes it useable for understanding +them or anything else actually. It tends now to describe the very same +algorithm under 50 different names and pages with few understandable +by even people who fully understand the algorithm and the theory behind. + +Problem description: (that is what it can solve) +assume we have a 2d table, or you could call it a graph or matrix if you +prefer + + O O O O O O O + + O O O O O O O + + O O O O O O O + + O O O O O O O + + +That table has edges connecting points from each column to the next column +and each edge has a score like: (only some edge and scores shown to keep it +readable) + + + O--5--O-----O-----O-----O-----O + 2 / 7 / \ / \ / \ / + \ / \ / \ / \ / \ / + O7-/--O--/--O--/--O--/--O--/--O + \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/ + /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\ + O3-/--O--/--O--/--O--/--O--/--O + / \ / \ / \ / \ / \ + 1 \ 9 \ / \ / \ / \ + O--2--O--1--O--5--O--3--O--8--O + + + +Our goal is to find a path from left to right through it which +minimizes the sum of the score of all edges. +(and of course left/right is just a convention here it could be top down too) +Similarly the minimum could be the maximum by just fliping the sign, +Example of a path with scores: + + O O O O O O O + +>---O. O O .O-2-O O O + 5. .7 . + O O-1-O O O 8 O O + . + O O O O O O-1-O---> (sum here is 24) + + +The viterbi algorthm now solves this simply column by column +For the previous column each point has a best path and a associated +score: + + O-----5 O + \ + \ + O \ 1 O + \/ + /\ + O / 2 O + / + / + O-----2 O + + +To move one column forward we just need to find the best path and associated +scores for the next column +here are some edges we could choose from: + + + O-----5--3--O + \ \8 + \ \ + O \ 1--9--O + \/ \3 + /\ \ + O / 2--1--O + / \2 + / \ + O-----2--4--O + +Finding the new best pathes and scores for each point of our new column is +trivial given we know the previous column best pathes and scores: + + O-----0-----8 + \ + \ + O \ 0----10 + \/ + /\ + O / 0-----3 + / \ + / \ + O 0 4 + + +the viterbi algorthm continues exactly like this column for column until the +end and then just picks the path with the best score (above that would be the +one with score 3) + + +Author: Michael niedermayer +Copyright LGPL + |