doc: remove doc/ffmpeg-mt-authorship.txt for release/0.10

we dont carry the whole git history in releases so theres no
point in having this in them either.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

.gitignore

@@ -1,68 +1,56 @@
-*.a
+.config
+.version
 *.o
 *.d
-*.def
-*.dll
-*.dylib
 *.exe
-*.exp
-*.h.c
-*.ilk
-*.lib
-*.pc
-*.pdb
-*.so
-*.so.*
-*.ver
+*.ho
 *-example
 *-test
 *_g
-/.config
-/.version
-/ffmpeg
-/ffplay
-/ffprobe
-/ffserver
-/config.*
-/version.h
-/doc/*.1
-/doc/*.3
-/doc/*.html
-/doc/*.pod
-/doc/avoptions_codec.texi
-/doc/avoptions_format.texi
-/doc/examples/decoding_encoding
-/doc/examples/demuxing
-/doc/examples/filtering_audio
-/doc/examples/filtering_video
-/doc/examples/metadata
-/doc/examples/muxing
-/doc/examples/resampling_audio
-/doc/examples/scaling_video
-/doc/fate.txt
-/doc/doxy/html/
-/doc/print_options
-/libavcodec/*_tablegen
-/libavcodec/*_tables.c
-/libavcodec/*_tables.h
-/libavutil/avconfig.h
-/tests/audiogen
-/tests/base64
-/tests/data/
-/tests/rotozoom
-/tests/tiny_psnr
-/tests/videogen
-/tests/vsynth1/
-/tools/aviocat
-/tools/ffbisect
-/tools/bisect.need
-/tools/cws2fws
-/tools/fourcc2pixfmt
-/tools/ffescape
-/tools/ffeval
-/tools/graph2dot
-/tools/ismindex
-/tools/pktdumper
-/tools/probetest
-/tools/qt-faststart
-/tools/trasher
+*.def
+*.dll
+*.lib
+*.exp
+config.*
+doc/*.1
+doc/*.html
+doc/*.pod
+doc/fate.txt
+doxy
+ffmpeg
+ffplay
+ffprobe
+ffserver
+avconv
+libavcodec/*_tablegen
+libavcodec/*_tables.c
+libavcodec/*_tables.h
+libavcodec/codec_names.h
+libavcodec/libavcodec*
+libavcore/libavcore*
+libavdevice/libavdevice*
+libavfilter/libavfilter*
+libavformat/libavformat*
+libavutil/avconfig.h
+libavutil/libavutil*
+libpostproc/libpostproc*
+libswresample/libswresample*
+libswscale/libswscale*
+tests/audiogen
+tests/base64
+tests/data
+tests/rotozoom
+tests/tiny_psnr
+tests/videogen
+tests/vsynth1
+tests/vsynth2
+tools/aviocat
+tools/cws2fws
+tools/graph2dot
+tools/ismindex
+tools/lavfi-showfiltfmts
+tools/pktdumper
+tools/probetest
+tools/qt-faststart
+tools/trasher
+version.h

COPYING.LGPLv2.1

@@ -500,3 +500,5 @@ necessary.  Here is a sample; alter the names:
   Ty Coon, President of Vice
 
 That's all there is to it!
+
+

Changelog

@@ -0,0 +1,881 @@
+Entries are sorted chronologically from oldest to youngest within each release,
+releases are sorted from youngest to oldest.
+
+version next:
+
+version 0.10:
+- Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
+         CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
+         CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
+         CVE-2011-3950, CVE-2011-3951, CVE-2011-3952
+- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
+- SBaGen (SBG) binaural beats script demuxer
+- OpenMG Audio muxer
+- Timecode extraction in DV and MOV
+- thumbnail video filter
+- XML output in ffprobe
+- asplit audio filter
+- tinterlace video filter
+- astreamsync audio filter
+- amerge audio filter
+- ISMV (Smooth Streaming) muxer
+- GSM audio parser
+- SMJPEG muxer
+- XWD encoder and decoder
+- Automatic thread count based on detection number of (available) CPU cores
+- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
+- ffprobe -show_error option
+- Avid 1:1 10-bit RGB Packer codec
+- v308 Quicktime Uncompressed 4:4:4 encoder and decoder
+- yuv4 libquicktime packed 4:2:0 encoder and decoder
+- ffprobe -show_frames option
+- silencedetect audio filter
+- ffprobe -show_program_version, -show_library_versions, -show_versions options
+- rv34: frame-level multi-threading
+- optimized iMDCT transform on x86 using SSE for for mpegaudiodec
+- Improved PGS subtitle decoder
+- dumpgraph option to lavfi device
+- r210 and r10k encoders
+- ffwavesynth decoder
+- aviocat tool
+- ffeval tool
+
+
+version 0.9:
+
+- openal input device added
+- boxblur filter added
+- BWF muxer
+- Flash Screen Video 2 decoder
+- lavfi input device added
+- added avconv, which is almost the same for now, except
+for a few incompatible changes in the options, which will hopefully make them
+easier to use. The changes are:
+    * The options placement is now strictly enforced! While in theory the
+      options for ffmpeg should be given in [input options] -i INPUT [output
+      options] OUTPUT order, in practice it was possible to give output options
+      before the -i and it mostly worked. Except when it didn't - the behavior was
+      a bit inconsistent. In avconv, it is not possible to mix input and output
+      options. All non-global options are reset after an input or output filename.
+    * All per-file options are now truly per-file - they apply only to the next
+      input or output file and specifying different values for different files
+      will now work properly (notably -ss and -t options).
+    * All per-stream options are now truly per-stream - it is possible to
+      specify which stream(s) should a given option apply to. See the Stream
+      specifiers section in the avconv manual for details.
+    * In ffmpeg some options (like -newvideo/-newaudio/...) are irregular in the
+      sense that they're specified after the output filename instead of before,
+      like all other options. In avconv this irregularity is removed, all options
+      apply to the next input or output file.
+    * -newvideo/-newaudio/-newsubtitle options were removed. Not only were they
+      irregular and highly confusing, they were also redundant. In avconv the -map
+      option will create new streams in the output file and map input streams to
+      them. E.g. avconv -i INPUT -map 0 OUTPUT will create an output stream for
+      each stream in the first input file.
+    * The -map option now has slightly different and more powerful syntax:
+        + Colons (':') are used to separate file index/stream type/stream index
+          instead of dots. Comma (',') is used to separate the sync stream instead
+          of colon.. This is done for consistency with other options.
+        + It's possible to specify stream type. E.g. -map 0:a:2 creates an
+          output stream from the third input audio stream.
+        + Omitting the stream index now maps all the streams of the given type,
+          not just the first. E.g. -map 0:s creates output streams for all the
+          subtitle streams in the first input file.
+        + Since -map can now match multiple streams, negative mappings were
+          introduced. Negative mappings disable some streams from an already
+          defined map. E.g. '-map 0 -map -0:a:1' means 'create output streams for
+          all the stream in the first input file, except for the second audio
+          stream'.
+    * There is a new option -c (or -codec) for choosing the decoder/encoder to
+      use, which allows to precisely specify target stream(s) consistently with
+      other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
+      libvorbis sets the codec for the first audio stream and -c copy copies all
+      the streams without reencoding. Old -vcodec/-acodec/-scodec options are now
+      aliases to -c:v/a/s
+    * It is now possible to precisely specify which stream should an AVOption
+      apply to. E.g. -b:v:0 2M sets the bitrate for the first video stream, while
+      -b:a 128k sets the bitrate for all audio streams. Note that the old -ab 128k
+      syntax is deprecated and will stop working soon.
+    * -map_chapters now takes only an input file index and applies to the next
+      output file. This is consistent with how all the other options work.
+    * -map_metadata now takes only an input metadata specifier and applies to
+      the next output file. Output metadata specifier is now part of the option
+      name, similarly to the AVOptions/map/codec feature above.
+    * -metadata can now be used to set metadata on streams and chapters, e.g.
+      -metadata:s:1 language=eng sets the language of the first stream to 'eng'.
+      This made -vlang/-alang/-slang options redundant, so they were removed.
+    * -qscale option now uses stream specifiers and applies to all streams, not
+      just video. I.e. plain -qscale number would now apply to all streams. To get
+      the old behavior, use -qscale:v. Also there is now a shortcut -q for -qscale
+      and -aq is now an alias for -q:a.
+    * -vbsf/-absf/-sbsf options were removed and replaced by a -bsf option which
+      uses stream specifiers. Use -bsf:v/a/s instead of the old options.
+    * -itsscale option now uses stream specifiers, so its argument is only the
+      scale parameter.
+    * -intra option was removed, use -g 0 for the same effect.
+    * -psnr option was removed, use -flags +psnr for the same effect.
+    * -vf option is now an alias to the new -filter option, which uses stream specifiers.
+    * -vframes/-aframes/-dframes options are now aliases to the new -frames option.
+    * -vtag/-atag/-stag options are now aliases to the new -tag option.
+- XMV demuxer
+- LOAS demuxer
+- ashowinfo filter added
+- Windows Media Image decoder
+- amovie source added
+- LATM muxer/demuxer
+- Speex encoder via libspeex
+- JSON output in ffprobe
+- WTV muxer
+- Optional C++ Support (needed for libstagefright)
+- H.264 Decoding on Android via Stagefright
+- Prores decoder
+- BIN/XBIN/ADF/IDF text file decoder
+- aconvert audio filter added
+- audio support to lavfi input device added
+- libcdio-paranoia input device for audio CD grabbing
+- Apple ProRes decoder
+- CELT in Ogg demuxing
+- G.723.1 demuxer and decoder
+- libmodplug support (--enable-libmodplug)
+- VC-1 interlaced decoding
+- libutvideo wrapper (--enable-libutvideo)
+- aevalsrc audio source added
+- Ut Video decoder
+- Speex encoding via libspeex
+- 4:2:2 H.264 decoding support
+- 4:2:2 and 4:4:4 H.264 encoding with libx264
+- Pulseaudio input device
+- Prores encoder
+- Video Decoder Acceleration (VDA) HWAccel module.
+- replacement Indeo 3 decoder
+- new ffmpeg option: -map_channel
+- volume audio filter added
+- earwax audio filter added
+- libv4l2 support (--enable-libv4l2)
+- TLS/SSL and HTTPS protocol support
+- AVOptions API rewritten and documented
+- most of CODEC_FLAG2_*, some CODEC_FLAG_* and many codec-specific fields in
+  AVCodecContext deprecated. Codec private options should be used instead.
+- Properly working defaults in libx264 wrapper, support for native presets.
+- Encrypted OMA files support
+- Discworld II BMV decoding support
+- VBLE Decoder
+- OS X Video Decoder Acceleration (VDA) support
+- compact and csv output in ffprobe
+- pan audio filter
+- IFF Amiga Continuous Bitmap (ACBM) decoder
+- ass filter
+- CRI ADX audio format muxer and demuxer
+- Playstation Portable PMP format demuxer
+- Microsoft Windows ICO demuxer
+- life source
+- PCM format support in OMA demuxer
+- CLJR encoder
+- new option: -report
+- Dxtory capture format decoder
+- cellauto source
+- Simple segmenting muxer
+- Indeo 4 decoder
+- SMJPEG demuxer
+
+
+version 0.8:
+
+- many many things we forgot because we rather write code than changelogs
+- WebM support in Matroska de/muxer
+- low overhead Ogg muxing
+- MMS-TCP support
+- VP8 de/encoding via libvpx
+- Demuxer for On2's IVF format
+- Pictor/PC Paint decoder
+- HE-AAC v2 decoder
+- HE-AAC v2 encoding with libaacplus
+- libfaad2 wrapper removed
+- DTS-ES extension (XCh) decoding support
+- native VP8 decoder
+- RTSP tunneling over HTTP
+- RTP depacketization of SVQ3
+- -strict inofficial replaced by -strict unofficial
+- ffplay -exitonkeydown and -exitonmousedown options added
+- native GSM / GSM MS decoder
+- RTP depacketization of QDM2
+- ANSI/ASCII art playback system
+- Lego Mindstorms RSO de/muxer
+- libavcore added (and subsequently removed)
+- SubRip subtitle file muxer and demuxer
+- Chinese AVS encoding via libxavs
+- ffprobe -show_packets option added
+- RTP packetization of Theora and Vorbis
+- RTP depacketization of MP4A-LATM
+- RTP packetization and depacketization of VP8
+- hflip filter
+- Apple HTTP Live Streaming demuxer
+- a64 codec
+- MMS-HTTP support
+- G.722 ADPCM audio encoder/decoder
+- R10k video decoder
+- ocv_smooth filter
+- frei0r wrapper filter
+- change crop filter syntax to width:height:x:y
+- make the crop filter accept parametric expressions
+- make ffprobe accept AVFormatContext options
+- yadif filter
+- blackframe filter
+- Demuxer for Leitch/Harris' VR native stream format (LXF)
+- RTP depacketization of the X-QT QuickTime format
+- SAP (Session Announcement Protocol, RFC 2974) muxer and demuxer
+- cropdetect filter
+- ffmpeg -crop* options removed
+- transpose filter added
+- ffmpeg -force_key_frames option added
+- demuxer for receiving raw rtp:// URLs without an SDP description
+- single stream LATM/LOAS decoder
+- setpts filter added
+- Win64 support for optimized x86 assembly functions
+- MJPEG/AVI1 to JPEG/JFIF bitstream filter
+- ASS subtitle encoder and decoder
+- IEC 61937 encapsulation for E-AC-3, TrueHD, DTS-HD (for HDMI passthrough)
+- overlay filter added
+- rename aspect filter to setdar, and pixelaspect to setsar
+- IEC 61937 demuxer
+- Mobotix .mxg demuxer
+- frei0r source added
+- hqdn3d filter added
+- RTP depacketization of QCELP
+- FLAC parser added
+- gradfun filter added
+- AMR-WB decoder
+- replace the ocv_smooth filter with a more generic ocv filter
+- Windows Televison (WTV) demuxer
+- FFmpeg metadata format muxer and demuxer
+- SubRip (srt) subtitle encoder and decoder
+- floating-point AC-3 encoder added
+- Lagarith decoder
+- ffmpeg -copytb option added
+- IVF muxer added
+- Wing Commander IV movies decoder added
+- movie source added
+- Bink version 'b' audio and video decoder
+- Bitmap Brothers JV playback system
+- Apple HTTP Live Streaming protocol handler
+- sndio support for playback and record
+- Linux framebuffer input device added
+- Chronomaster DFA decoder
+- DPX image encoder
+- MicroDVD subtitle file muxer and demuxer
+- Playstation Portable PMP format demuxer
+- fieldorder video filter added
+- AAC encoding via libvo-aacenc
+- AMR-WB encoding via libvo-amrwbenc
+- xWMA demuxer
+- Mobotix MxPEG decoder
+- VP8 frame-multithreading
+- NEON optimizations for VP8
+- Lots of deprecated API cruft removed
+- fft and imdct optimizations for AVX (Sandy Bridge) processors
+- showinfo filter added
+- SMPTE 302M AES3 audio decoder
+- Apple Core Audio Format muxer
+- 9bit and 10bit per sample support in the H.264 decoder
+- 9bit and 10bit FFV1 encoding / decoding
+- split filter added
+- select filter added
+- sdl output device added
+- libmpcodecs video filter support (3 times as many filters than before)
+- mpeg2 aspect ratio dection fixed
+- libxvid aspect pickiness fixed
+- Frame multithreaded decoding
+- E-AC-3 audio encoder
+- ac3enc: add channel coupling support
+- floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
+- H264/MPEG frame-level multi-threading
+- All av_metadata_* functions renamed to av_dict_* and moved to libavutil
+- 4:4:4 H.264 decoding support
+- 10-bit H.264 optimizations for x86
+- lut, lutrgb, and lutyuv filters added
+- buffersink libavfilter sink added
+- Bump libswscale for recently reported ABI break
+- New J2K encoder (via OpenJPEG)
+
+
+version 0.7:
+
+- all the changes for 0.8, but keeping API/ABI compatibility with the 0.6 release
+
+
+version 0.6:
+
+- PB-frame decoding for H.263
+- deprecated vhook subsystem removed
+- deprecated old scaler removed
+- VQF demuxer
+- Alpha channel scaler
+- PCX encoder
+- RTP packetization of H.263
+- RTP packetization of AMR
+- RTP depacketization of Vorbis
+- CorePNG decoding support
+- Cook multichannel decoding support
+- introduced avlanguage helpers in libavformat
+- 8088flex TMV demuxer and decoder
+- per-stream language-tags extraction in asfdec
+- V210 decoder and encoder
+- remaining GPL parts in AC-3 decoder converted to LGPL
+- QCP demuxer
+- SoX native format muxer and demuxer
+- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
+- DPX image decoder
+- Electronic Arts Madcow decoder
+- DivX (XSUB) subtitle encoder
+- nonfree libamr support for AMR-NB/WB decoding/encoding removed
+- experimental AAC encoder
+- RTP depacketization of ASF and RTSP from WMS servers
+- RTMP support in libavformat
+- noX handling for OPT_BOOL X options
+- Wave64 demuxer
+- IEC-61937 compatible Muxer
+- TwinVQ decoder
+- Bluray (PGS) subtitle decoder
+- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
+- WMA Pro decoder
+- Core Audio Format demuxer
+- Atrac1 decoder
+- MD STUDIO audio demuxer
+- RF64 support in WAV demuxer
+- MPEG-4 Audio Lossless Coding (ALS) decoder
+- -formats option split into -formats, -codecs, -bsfs, and -protocols
+- IV8 demuxer
+- CDG demuxer and decoder
+- R210 decoder
+- Auravision Aura 1 and 2 decoders
+- Deluxe Paint Animation playback system
+- SIPR decoder
+- Adobe Filmstrip muxer and demuxer
+- RTP depacketization of H.263
+- Bink demuxer and audio/video decoders
+- enable symbol versioning by default for linkers that support it
+- IFF PBM/ILBM bitmap decoder
+- concat protocol
+- Indeo 5 decoder
+- RTP depacketization of AMR
+- WMA Voice decoder
+- ffprobe tool
+- AMR-NB decoder
+- RTSP muxer
+- HE-AAC v1 decoder
+- Kega Game Video (KGV1) decoder
+- VorbisComment writing for FLAC, Ogg FLAC and Ogg Speex files
+- RTP depacketization of Theora
+- HTTP Digest authentication
+- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
+- Psygnosis YOP demuxer and video decoder
+- spectral extension support in the E-AC-3 decoder
+- unsharp video filter
+- RTP hinting in the mov/3gp/mp4 muxer
+- Dirac in Ogg demuxing
+- seek to keyframes in Ogg
+- 4:2:2 and 4:4:4 Theora decoding
+- 35% faster VP3/Theora decoding
+- faster AAC decoding
+- faster H.264 decoding
+- RealAudio 1.0 (14.4K) encoder
+
+
+version 0.5:
+
+- DV50 AKA DVCPRO50 encoder, decoder, muxer and demuxer
+- TechSmith Camtasia (TSCC) video decoder
+- IBM Ultimotion (ULTI) video decoder
+- Sierra Online audio file demuxer and decoder
+- Apple QuickDraw (qdrw) video decoder
+- Creative ADPCM audio decoder (16 bits as well as 8 bits schemes)
+- Electronic Arts Multimedia (WVE/UV2/etc.) file demuxer
+- Miro VideoXL (VIXL) video decoder
+- H.261 video encoder
+- QPEG video decoder
+- Nullsoft Video (NSV) file demuxer
+- Shorten audio decoder
+- LOCO video decoder
+- Apple Lossless Audio Codec (ALAC) decoder
+- Winnov WNV1 video decoder
+- Autodesk Animator Studio Codec (AASC) decoder
+- Indeo 2 video decoder
+- Fraps FPS1 video decoder
+- Snow video encoder/decoder
+- Sonic audio encoder/decoder
+- Vorbis audio decoder
+- Macromedia ADPCM decoder
+- Duck TrueMotion 2 video decoder
+- support for decoding FLX and DTA extensions in FLIC files
+- H.264 custom quantization matrices support
+- ffserver fixed, it should now be usable again
+- QDM2 audio decoder
+- Real Cooker audio decoder
+- TrueSpeech audio decoder
+- WMA2 audio decoder fixed, now all files should play correctly
+- RealAudio 14.4 and 28.8 decoders fixed
+- JPEG-LS decoder
+- build system improvements
+- tabs and trailing whitespace removed from the codebase
+- CamStudio video decoder
+- AIFF/AIFF-C audio format, encoding and decoding
+- ADTS AAC file reading and writing
+- Creative VOC file reading and writing
+- American Laser Games multimedia (*.mm) playback system
+- Zip Motion Blocks Video decoder
+- improved Theora/VP3 decoder
+- True Audio (TTA) decoder
+- AVS demuxer and video decoder
+- JPEG-LS encoder
+- Smacker demuxer and decoder
+- NuppelVideo/MythTV demuxer and RTjpeg decoder
+- KMVC decoder
+- MPEG-2 intra VLC support
+- MPEG-2 4:2:2 encoder
+- Flash Screen Video decoder
+- GXF demuxer
+- Chinese AVS decoder
+- GXF muxer
+- MXF demuxer
+- VC-1/WMV3/WMV9 video decoder
+- MacIntel support
+- AVISynth support
+- VMware video decoder
+- VP5 video decoder
+- VP6 video decoder
+- WavPack lossless audio decoder
+- Targa (.TGA) picture decoder
+- Vorbis audio encoder
+- Delphine Software .cin demuxer/audio and video decoder
+- Tiertex .seq demuxer/video decoder
+- MTV demuxer
+- TIFF picture encoder and decoder
+- GIF picture decoder
+- Intel Music Coder decoder
+- Zip Motion Blocks Video encoder
+- Musepack decoder
+- Flash Screen Video encoder
+- Theora encoding via libtheora
+- BMP encoder
+- WMA encoder
+- GSM-MS encoder and decoder
+- DCA decoder
+- DXA demuxer and decoder
+- DNxHD decoder
+- Gamecube movie (.THP) playback system
+- Blackfin optimizations
+- Interplay C93 demuxer and video decoder
+- Bethsoft VID demuxer and video decoder
+- CRYO APC demuxer
+- Atrac3 decoder
+- V.Flash PTX decoder
+- RoQ muxer, RoQ audio encoder
+- Renderware TXD demuxer and decoder
+- extern C declarations for C++ removed from headers
+- sws_flags command line option
+- codebook generator
+- RoQ video encoder
+- QTRLE encoder
+- OS/2 support removed and restored again
+- AC-3 decoder
+- NUT muxer
+- additional SPARC (VIS) optimizations
+- Matroska muxer
+- slice-based parallel H.264 decoding
+- Monkey's Audio demuxer and decoder
+- AMV audio and video decoder
+- DNxHD encoder
+- H.264 PAFF decoding
+- Nellymoser ASAO decoder
+- Beam Software SIFF demuxer and decoder
+- libvorbis Vorbis decoding removed in favor of native decoder
+- IntraX8 (J-Frame) subdecoder for WMV2 and VC-1
+- Ogg (Theora, Vorbis and FLAC) muxer
+- The "device" muxers and demuxers are now in a new libavdevice library
+- PC Paintbrush PCX decoder
+- Sun Rasterfile decoder
+- TechnoTrend PVA demuxer
+- Linux Media Labs MPEG-4 (LMLM4) demuxer
+- AVM2 (Flash 9) SWF muxer
+- QT variant of IMA ADPCM encoder
+- VFW grabber
+- iPod/iPhone compatible mp4 muxer
+- Mimic decoder
+- MSN TCP Webcam stream demuxer
+- RL2 demuxer / decoder
+- IFF demuxer
+- 8SVX audio decoder
+- non-recursive Makefiles
+- BFI demuxer
+- MAXIS EA XA (.xa) demuxer / decoder
+- BFI video decoder
+- OMA demuxer
+- MLP/TrueHD decoder
+- Electronic Arts CMV decoder
+- Motion Pixels Video decoder
+- Motion Pixels MVI demuxer
+- removed animated GIF decoder/demuxer
+- D-Cinema audio muxer
+- Electronic Arts TGV decoder
+- Apple Lossless Audio Codec (ALAC) encoder
+- AAC decoder
+- floating point PCM encoder/decoder
+- MXF muxer
+- DV100 AKA DVCPRO HD decoder and demuxer
+- E-AC-3 support added to AC-3 decoder
+- Nellymoser ASAO encoder
+- ASS and SSA demuxer and muxer
+- liba52 wrapper removed
+- SVQ3 watermark decoding support
+- Speex decoding via libspeex
+- Electronic Arts TGQ decoder
+- RV40 decoder
+- QCELP / PureVoice decoder
+- RV30 decoder
+- hybrid WavPack support
+- R3D REDCODE demuxer
+- ALSA support for playback and record
+- Electronic Arts TQI decoder
+- OpenJPEG based JPEG 2000 decoder
+- NC (NC4600) camera file demuxer
+- Gopher client support
+- MXF D-10 muxer
+- generic metadata API
+- flash ScreenVideo2 encoder
+
+
+version 0.4.9-pre1:
+
+- DV encoder, DV muxer
+- Microsoft RLE video decoder
+- Microsoft Video-1 decoder
+- Apple Animation (RLE) decoder
+- Apple Graphics (SMC) decoder
+- Apple Video (RPZA) decoder
+- Cinepak decoder
+- Sega FILM (CPK) file demuxer
+- Westwood multimedia support (VQA & AUD files)
+- Id Quake II CIN playback support
+- 8BPS video decoder
+- FLIC playback support
+- RealVideo 2.0 (RV20) decoder
+- Duck TrueMotion v1 (DUCK) video decoder
+- Sierra VMD demuxer and video decoder
+- MSZH and ZLIB decoder support
+- SVQ1 video encoder
+- AMR-WB support
+- PPC optimizations
+- rate distortion optimal cbp support
+- rate distorted optimal ac prediction for MPEG-4
+- rate distorted optimal lambda->qp support
+- AAC encoding with libfaac
+- Sunplus JPEG codec (SP5X) support
+- use Lagrange multipler instead of QP for ratecontrol
+- Theora/VP3 decoding support
+- XA and ADX ADPCM codecs
+- export MPEG-2 active display area / pan scan
+- Add support for configuring with IBM XLC
+- floating point AAN DCT
+- initial support for zygo video (not complete)
+- RGB ffv1 support
+- new audio/video parser API
+- av_log() system
+- av_read_frame() and av_seek_frame() support
+- missing last frame fixes
+- seek by mouse in ffplay
+- noise reduction of DCT coefficients
+- H.263 OBMC & 4MV support
+- H.263 alternative inter vlc support
+- H.263 loop filter
+- H.263 slice structured mode
+- interlaced DCT support for MPEG-2 encoding
+- stuffing to stay above min_bitrate
+- MB type & QP visualization
+- frame stepping for ffplay
+- interlaced motion estimation
+- alternate scantable support
+- SVCD scan offset support
+- closed GOP support
+- SSE2 FDCT
+- quantizer noise shaping
+- G.726 ADPCM audio codec
+- MS ADPCM encoding
+- multithreaded/SMP motion estimation
+- multithreaded/SMP encoding for MPEG-1/MPEG-2/MPEG-4/H.263
+- multithreaded/SMP decoding for MPEG-2
+- FLAC decoder
+- Metrowerks CodeWarrior suppport
+- H.263+ custom pcf support
+- nicer output for 'ffmpeg -formats'
+- Matroska demuxer
+- SGI image format, encoding and decoding
+- H.264 loop filter support
+- H.264 CABAC support
+- nicer looking arrows for the motion vector visualization
+- improved VCD support
+- audio timestamp drift compensation
+- MPEG-2 YUV 422/444 support
+- polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample
+- better image scaling
+- H.261 support
+- correctly interleave packets during encoding
+- VIS optimized motion compensation
+- intra_dc_precision>0 encoding support
+- support reuse of motion vectors/MB types/field select values of the source video
+- more accurate deblock filter
+- padding support
+- many optimizations and bugfixes
+- FunCom ISS audio file demuxer and according ADPCM decoding
+
+
+version 0.4.8:
+
+- MPEG-2 video encoding (Michael)
+- Id RoQ playback subsystem (Mike Melanson and Tim Ferguson)
+- Wing Commander III Movie (.mve) file playback subsystem (Mike Melanson
+  and Mario Brito)
+- Xan DPCM audio decoder (Mario Brito)
+- Interplay MVE playback subsystem (Mike Melanson)
+- Duck DK3 and DK4 ADPCM audio decoders (Mike Melanson)
+
+
+version 0.4.7:
+
+- RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from mplayerhq
+  (originally from public domain player for Amiga at http://www.honeypot.net/audio)
+- current version now also compiles with older GCC (Fabrice)
+- 4X multimedia playback system including 4xm file demuxer (Mike
+  Melanson), and 4X video and audio codecs (Michael)
+- Creative YUV (CYUV) decoder (Mike Melanson)
+- FFV1 codec (our very simple lossless intra only codec, compresses much better
+  than HuffYUV) (Michael)
+- ASV1 (Asus), H.264, Intel indeo3 codecs have been added (various)
+- tiny PNG encoder and decoder, tiny GIF decoder, PAM decoder (PPM with
+  alpha support), JPEG YUV colorspace support. (Fabrice Bellard)
+- ffplay has been replaced with a newer version which uses SDL (optionally)
+  for multiplatform support (Fabrice)
+- Sorenson Version 3 codec (SVQ3) support has been added (decoding only) - donated
+  by anonymous
+- AMR format has been added (Johannes Carlsson)
+- 3GP support has been added (Johannes Carlsson)
+- VP3 codec has been added (Mike Melanson)
+- more MPEG-1/2 fixes
+- better multiplatform support, MS Visual Studio fixes (various)
+- AltiVec optimizations (Magnus Damn and others)
+- SH4 processor support has been added (BERO)
+- new public interfaces (avcodec_get_pix_fmt) (Roman Shaposhnick)
+- VOB streaming support (Brian Foley)
+- better MP3 autodetection (Andriy Rysin)
+- qpel encoding (Michael)
+- 4mv+b frames encoding finally fixed (Michael)
+- chroma ME (Michael)
+- 5 comparison functions for ME (Michael)
+- B-frame encoding speedup (Michael)
+- WMV2 codec (unfinished - Michael)
+- user specified diamond size for EPZS (Michael)
+- Playstation STR playback subsystem, still experimental (Mike and Michael)
+- ASV2 codec (Michael)
+- CLJR decoder (Alex)
+
+.. And lots more new enhancements and fixes.
+
+
+version 0.4.6:
+
+- completely new integer only MPEG audio layer 1/2/3 decoder rewritten
+  from scratch
+- Recoded DCT and motion vector search with gcc (no longer depends on nasm)
+- fix quantization bug in AC3 encoder
+- added PCM codecs and format. Corrected WAV/AVI/ASF PCM issues
+- added prototype ffplay program
+- added GOB header parsing on H.263/H.263+ decoder (Juanjo)
+- bug fix on MCBPC tables of H.263 (Juanjo)
+- bug fix on DC coefficients of H.263 (Juanjo)
+- added Advanced Prediction Mode on H.263/H.263+ decoder (Juanjo)
+- now we can decode H.263 streams found in QuickTime files (Juanjo)
+- now we can decode H.263 streams found in VIVO v1 files(Juanjo)
+- preliminary RTP "friendly" mode for H.263/H.263+ coding. (Juanjo)
+- added GOB header for H.263/H.263+ coding on RTP mode (Juanjo)
+- now H.263 picture size is returned on the first decoded frame (Juanjo)
+- added first regression tests
+- added MPEG-2 TS demuxer
+- new demux API for libav
+- more accurate and faster IDCT (Michael)
+- faster and entropy-controlled motion search (Michael)
+- two pass video encoding (Michael)
+- new video rate control (Michael)
+- added MSMPEG4V1, MSMPEGV2 and WMV1 support (Michael)
+- great performance improvement of video encoders and decoders (Michael)
+- new and faster bit readers and vlc parsers (Michael)
+- high quality encoding mode: tries all macroblock/VLC types (Michael)
+- added DV video decoder
+- preliminary RTP/RTSP support in ffserver and libavformat
+- H.263+ AIC decoding/encoding support (Juanjo)
+- VCD MPEG-PS mode (Juanjo)
+- PSNR stuff (Juanjo)
+- simple stats output (Juanjo)
+- 16-bit and 15-bit RGB/BGR/GBR support (Bisqwit)
+
+
+version 0.4.5:
+
+- some header fixes (Zdenek Kabelac <kabi at informatics.muni.cz>)
+- many MMX optimizations (Nick Kurshev <nickols_k at mail.ru>)
+- added configure system (actually a small shell script)
+- added MPEG audio layer 1/2/3 decoding using LGPL'ed mpglib by
+  Michael Hipp (temporary solution - waiting for integer only
+  decoder)
+- fixed VIDIOCSYNC interrupt
+- added Intel H.263 decoding support ('I263' AVI fourCC)
+- added Real Video 1.0 decoding (needs further testing)
+- simplified image formats again. Added PGM format (=grey
+  pgm). Renamed old PGM to PGMYUV.
+- fixed msmpeg4 slice issues (tell me if you still find problems)
+- fixed OpenDivX bugs with newer versions (added VOL header decoding)
+- added support for MPlayer interface
+- added macroblock skip optimization
+- added MJPEG decoder
+- added mmx/mmxext IDCT from libmpeg2
+- added pgmyuvpipe, ppm, and ppm_pipe formats (original patch by Celer
+  <celer at shell.scrypt.net>)
+- added pixel format conversion layer (e.g. for MJPEG or PPM)
+- added deinterlacing option
+- MPEG-1/2 fixes
+- MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
+- ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
+- Windows porting of file converter
+- added MJPEG raw format (input/ouput)
+- added JPEG image format support (input/output)
+
+
+version 0.4.4:
+
+- fixed some std header definitions (Bjorn Lindgren
+  <bjorn.e.lindgren at telia.com>).
+- added MPEG demuxer (MPEG-1 and 2 compatible).
+- added ASF demuxer
+- added prototype RM demuxer
+- added AC3 decoding (done with libac3 by Aaron Holtzman)
+- added decoding codec parameter guessing (.e.g. for MPEG, because the
+  header does not include them)
+- fixed header generation in MPEG-1, AVI and ASF muxer: wmplayer can now
+  play them (only tested video)
+- fixed H.263 white bug
+- fixed phase rounding in img resample filter
+- add MMX code for polyphase img resample filter
+- added CPU autodetection
+- added generic title/author/copyright/comment string handling (ASF and RM
+  use them)
+- added SWF demux to extract MP3 track (not usable yet because no MP3
+  decoder)
+- added fractional frame rate support
+- codecs are no longer searched by read_header() (should fix ffserver
+  segfault)
+
+
+version 0.4.3:
+
+- BGR24 patch (initial patch by Jeroen Vreeken <pe1rxq at amsat.org>)
+- fixed raw yuv output
+- added motion rounding support in MPEG-4
+- fixed motion bug rounding in MSMPEG4
+- added B-frame handling in video core
+- added full MPEG-1 decoding support
+- added partial (frame only) MPEG-2 support
+- changed the FOURCC code for H.263 to "U263" to be able to see the
+  +AVI/H.263 file with the UB Video H.263+ decoder. MPlayer works with
+  this +codec ;) (JuanJo).
+- Halfpel motion estimation after MB type selection (JuanJo)
+- added pgm and .Y.U.V output format
+- suppressed 'img:' protocol. Simply use: /tmp/test%d.[pgm|Y] as input or
+  output.
+- added pgmpipe I/O format (original patch from Martin Aumueller
+  <lists at reserv.at>, but changed completely since we use a format
+  instead of a protocol)
+
+
+version 0.4.2:
+
+- added H.263/MPEG-4/MSMPEG4 decoding support. MPEG-4 decoding support
+  (for OpenDivX) is almost complete: 8x8 MVs and rounding are
+  missing. MSMPEG4 support is complete.
+- added prototype MPEG-1 decoder. Only I- and P-frames handled yet (it
+  can decode ffmpeg MPEGs :-)).
+- added libavcodec API documentation (see apiexample.c).
+- fixed image polyphase bug (the bottom of some images could be
+  greenish)
+- added support for non clipped motion vectors (decoding only)
+  and image sizes non-multiple of 16
+- added support for AC prediction (decoding only)
+- added file overwrite confirmation (can be disabled with -y)
+- added custom size picture to H.263 using H.263+ (Juanjo)
+
+
+version 0.4.1:
+
+- added MSMPEG4 (aka DivX) compatible encoder. Changed default codec
+  of AVI and ASF to DIV3.
+- added -me option to set motion estimation method
+  (default=log). suppressed redundant -hq option.
+- added options -acodec and -vcodec to force a given codec (useful for
+  AVI for example)
+- fixed -an option
+- improved dct_quantize speed
+- factorized some motion estimation code
+
+
+version 0.4.0:
+
+- removing grab code from ffserver and moved it to ffmpeg. Added
+  multistream support to ffmpeg.
+- added timeshifting support for live feeds (option ?date=xxx in the
+  URL)
+- added high quality image resize code with polyphase filter (need
+  mmx/see optimization). Enable multiple image size support in ffserver.
+- added multi live feed support in ffserver
+- suppressed master feature from ffserver (it should be done with an
+  external program which opens the .ffm url and writes it to another
+  ffserver)
+- added preliminary support for video stream parsing (WAV and AVI half
+  done). Added proper support for audio/video file conversion in
+  ffmpeg.
+- added preliminary support for video file sending from ffserver
+- redesigning I/O subsystem: now using URL based input and output
+  (see avio.h)
+- added WAV format support
+- added "tty user interface" to ffmpeg to stop grabbing gracefully
+- added MMX/SSE optimizations to SAD (Sums of Absolutes Differences)
+  (Juan J. Sierralta P. a.k.a. "Juanjo" <juanjo at atmlab.utfsm.cl>)
+- added MMX DCT from mpeg2_movie 1.5 (Juanjo)
+- added new motion estimation algorithms, log and phods (Juanjo)
+- changed directories: libav for format handling, libavcodec for
+  codecs
+
+
+version 0.3.4:
+
+- added stereo in MPEG audio encoder
+
+
+version 0.3.3:
+
+- added 'high quality' mode which use motion vectors. It can be used in
+  real time at low resolution.
+- fixed rounding problems which caused quality problems at high
+  bitrates and large GOP size
+
+
+version 0.3.2: small fixes
+
+- ASF fixes
+- put_seek bug fix
+
+
+version 0.3.1: added avi/divx support
+
+- added AVI support
+- added MPEG-4 codec compatible with OpenDivX. It is based on the H.263 codec
+- added sound for flash format (not tested)
+
+
+version 0.3: initial public release

Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 1.1.16
+PROJECT_NUMBER         = 0.10
 
 # With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
@@ -44,7 +44,7 @@ PROJECT_LOGO           =
 # If a relative path is entered, it will be relative to the location
 # where doxygen was started. If left blank the current directory will be used.
 
-OUTPUT_DIRECTORY       = doc/doxy
+OUTPUT_DIRECTORY       = doxy
 
 # If the CREATE_SUBDIRS tag is set to YES, then doxygen will create
 # 4096 sub-directories (in 2 levels) under the output directory of each output
@@ -288,7 +288,7 @@ TYPEDEF_HIDES_STRUCT   = NO
 # causing a significant performance penality.
 # If the system has enough physical memory increasing the cache will improve the
 # performance by keeping more symbols in memory. Note that the value works on
-# a logarithmic scale so increasing the size by one will roughly double the
+# a logarithmic scale so increasing the size by one will rougly double the
 # memory usage. The cache size is given by this formula:
 # 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
 # corresponding to a cache size of 2^16 = 65536 symbols
@@ -489,6 +489,12 @@ MAX_INITIALIZER_LINES  = 30
 
 SHOW_USED_FILES        = YES
 
+# If the sources in your project are distributed over multiple directories
+# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy
+# in the documentation. The default is NO.
+
+SHOW_DIRECTORIES       = NO
+
 # Set the SHOW_FILES tag to NO to disable the generation of the Files page.
 # This will remove the Files entry from the Quick Index and from the
 # Folder Tree View (if specified). The default is YES.
@@ -639,14 +645,15 @@ EXCLUDE_SYMBOLS        =
 # directories that contain example code fragments that are included (see
 # the \include command).
 
-EXAMPLE_PATH           = doc/examples/
+EXAMPLE_PATH           = libavcodec/ \
+                         libavformat/
 
 # If the value of the EXAMPLE_PATH tag contains directories, you can use the
 # EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
 # and *.h) to filter out the source-files in the directories. If left
 # blank all files are included.
 
-EXAMPLE_PATTERNS       = *.c
+EXAMPLE_PATTERNS       = *-example.c
 
 # If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be
 # searched for input files to be used with the \include or \dontinclude
@@ -793,13 +800,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.
 
-#HTML_HEADER            = doc/doxy/header.html
+HTML_HEADER            = doc/doxy/header.html
 
 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.
 
-#HTML_FOOTER            = doc/doxy/footer.html
+HTML_FOOTER            = doc/doxy/footer.html
 
 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -808,7 +815,7 @@ HTML_FILE_EXTENSION    = .html
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!
 
-#HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
+HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
 
 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
@@ -818,7 +825,7 @@ HTML_FILE_EXTENSION    = .html
 # 180 is cyan, 240 is blue, 300 purple, and 360 is red again.
 # The allowed range is 0 to 359.
 
-#HTML_COLORSTYLE_HUE    = 120
+HTML_COLORSTYLE_HUE    = 120
 
 # The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of
 # the colors in the HTML output. For a value of 0 the output will use
@@ -841,6 +848,12 @@ HTML_COLORSTYLE_GAMMA  = 80
 
 HTML_TIMESTAMP         = YES
 
+# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes,
+# files or namespaces will be aligned in HTML using tables. If set to
+# NO a bullet list will be used.
+
+HTML_ALIGN_MEMBERS     = YES
+
 # If the HTML_DYNAMIC_SECTIONS tag is set to YES then the generated HTML
 # documentation will contain sections that can be hidden and shown after the
 # page has loaded. For this to work a browser that supports
@@ -1021,6 +1034,11 @@ ENUM_VALUES_PER_LINE   = 4
 
 GENERATE_TREEVIEW      = NO
 
+# By enabling USE_INLINE_TREES, doxygen will generate the Groups, Directories,
+# and Class Hierarchy pages using a tree view instead of an ordered list.
+
+USE_INLINE_TREES       = NO
+
 # If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be
 # used to set the initial width (in pixels) of the frame in which the tree
 # is shown.
@@ -1356,9 +1374,14 @@ INCLUDE_FILE_PATTERNS  =
 # instead of the = operator.
 
 PREDEFINED             = "__attribute__(x)=" \
+                         "RENAME(x)=x ## _TMPL" \
+                         "DEF(x)=x ## _TMPL" \
+                         HAVE_AV_CONFIG_H \
+                         HAVE_MMX \
+                         HAVE_MMX2 \
+                         HAVE_AMD3DNOW \
                          "DECLARE_ALIGNED(a,t,n)=t n" \
-                         "offsetof(x,y)=0x42" \
-                         av_alloc_size \
+                         "offsetof(x,y)=0x42"
 
 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
 # this tag can be used to specify a list of macro names that should be expanded.

LICENSE

@@ -1,4 +1,5 @@
 FFmpeg:
+-------
 
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
 or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
@@ -13,44 +14,14 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are
 
 - libpostproc
-- libmpcodecs
 - optional x86 optimizations in the files
   libavcodec/x86/idct_mmx.c
-- libutvideo encoding/decoding wrappers in
-  libavcodec/libutvideo*.cpp
 - the X11 grabber in libavdevice/x11grab.c
-- the swresample test app in
-  libswresample/swresample-test.c
-- the texi2pod.pl tool
-- the following filters in libavfilter:
-    - f_ebur128.c
-    - vf_blackframe.c
-    - vf_boxblur.c
-    - vf_colormatrix.c
-    - vf_cropdetect.c
-    - vf_decimate.c
-    - vf_delogo.c
-    - vf_geq.c
-    - vf_histeq.c
-    - vf_hqdn3d.c
-    - vf_hue.c
-    - vf_kerndeint.c
-    - vf_mp.c
-    - vf_pp.c
-    - vf_smartblur.c
-    - vf_super2xsai.c
-    - vf_tinterlace.c
-    - vf_yadif.c
-    - vsrc_mptestsrc.c
 
 There are a handful of files under other licensing terms, namely:
 
-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
-  licensing details. Specifically note that you must credit the IJG in the
-  documentation accompanying your program if you only distribute executables.
-  You must also indicate any changes including additions and deletions to
-  those three files in the documentation.
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
+  are taken from libjpeg, see the top of the files for licensing details.
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
 the configure parameter --enable-version3 will activate this licensing option
@@ -58,32 +29,18 @@ for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
 COPYING.GPLv3 to learn the exact legal terms that apply in this case.
 
 
-external libraries
-==================
+external libraries:
+-------------------
 
-FFmpeg can be combined with a number of external libraries, which sometimes
-affect the licensing of binaries resulting from the combination.
+Some external libraries, e.g. libx264, are under GPL and can be used in
+conjunction with FFmpeg. They require --enable-gpl to be passed to configure
+as well.
 
-compatible libraries
---------------------
+The OpenCORE external libraries are under the Apache License 2.0. That license
+is incompatible with the LGPL v2.1 and the GPL v2, but not with version 3 of
+those licenses. So to combine the OpenCORE libraries with FFmpeg, the license
+version needs to be upgraded by passing --enable-version3 to configure.
 
-The libcdio, libx264, libxavs and libxvid libraries are under GPL. When
-combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing --enable-gpl to configure.
-
-The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
-license is incompatible with the LGPL v2.1 and the GPL v2, but not with
-version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing --enable-version3 to configure.
-
-incompatible libraries
-----------------------
-
-The Fraunhofer AAC library, FAAC and aacplus are under licenses which
-are incompatible with the GPLv2 and v3. We do not know for certain if their
-licenses are compatible with the LGPL.
-If you wish to enable these libraries, pass --enable-nonfree to configure.
-But note that if you enable any of these libraries the resulting binary will
-be under a complex license mix that is more restrictive than the LGPL and that
-may result in additional obligations. It is possible that these
-restrictions cause the resulting binary to be unredistributeable.
+The nonfree external libraries libfaac and libaacplus can be hooked up in FFmpeg.
+You need to pass --enable-nonfree to configure to enable it. Employ this option
+with care as FFmpeg then becomes nonfree and unredistributable.

MAINTAINERS

@@ -4,7 +4,7 @@ FFmpeg maintainers
 Below is a list of the people maintaining different parts of the
 FFmpeg code.
 
-Please try to keep entries where you are the maintainer up to date!
+Please try to keep entries where you are the maintainer upto date!
 
 Names in () mean that the maintainer currently has no time to maintain the code.
 A CC after the name means that the maintainer prefers to be CC-ed on patches
@@ -130,12 +130,9 @@ Codecs:
   bmp.c                                 Mans Rullgard, Kostya Shishkov
   cavs*                                 Stefan Gehrer
   celp_filters.*                        Vitor Sessak
-  cdxl.c                                Paul B Mahol
   cinepak.c                             Roberto Togni
   cljr                                  Alex Beregszaszi
-  cllc.c                                Derek Buitenhuis
   cook.c, cookdata.h                    Benjamin Larsson
-  cpia.c                                Stephan Hilb
   crystalhd.c                           Philip Langdale
   cscd.c                                Reimar Doeffinger
   dca.c                                 Kostya Shishkov, Benjamin Larsson
@@ -162,7 +159,6 @@ Codecs:
   indeo5*                               Kostya Shishkov
   interplayvideo.c                      Mike Melanson
   ivi*                                  Kostya Shishkov
-  jacosub*                              Clément Bœsch
   jpeg_ls.c                             Kostya Shishkov
   jvdec.c                               Peter Ross
   kmvc.c                                Kostya Shishkov
@@ -175,7 +171,6 @@ Codecs:
   libschroedinger*                      David Conrad
   libspeexdec.c                         Justin Ruggles
   libtheoraenc.c                        David Conrad
-  libutvideo*                           Derek Buitenhuis
   libvorbis.c                           David Conrad
   libxavs.c                             Stefan Gehrer
   libx264.c                             Mans Rullgard, Jason Garrett-Glaser
@@ -183,7 +178,7 @@ Codecs:
   lzo.h, lzo.c                          Reimar Doeffinger
   mdec.c                                Michael Niedermayer
   mimic.c                               Ramiro Polla
-  mjpeg*.c                              Michael Niedermayer
+  mjpeg.c                               Michael Niedermayer
   mlp*                                  Ramiro Polla
   mmvideo.c                             Peter Ross
   mpc*                                  Kostya Shishkov
@@ -194,7 +189,6 @@ Codecs:
   msvideo1.c                            Mike Melanson
   nellymoserdec.c                       Benjamin Larsson
   nuv.c                                 Reimar Doeffinger
-  paf.*                                 Paul B Mahol
   pcx.c                                 Ivo van Poorten
   pgssubdec.c                           Reimar Doeffinger
   ptx.c                                 Ivo van Poorten
@@ -219,7 +213,6 @@ Codecs:
   srt*                                  Aurelien Jacobs
   sunrast.c                             Ivo van Poorten
   svq3.c                                Michael Niedermayer
-  tak*                                  Paul B Mahol
   targa.c                               Kostya Shishkov
   tiff.c                                Kostya Shishkov
   truemotion1*                          Mike Melanson
@@ -234,8 +227,6 @@ Codecs:
   vble.c                                Derek Buitenhuis
   vc1*                                  Kostya Shishkov
   vcr1.c                                Michael Niedermayer
-  vda_h264_dec.c                        Xidorn Quan
-  vima.c                                Paul B Mahol
   vmnc.c                                Kostya Shishkov
   vorbis_enc.c                          Oded Shimon
   vorbis_dec.c                          Denes Balatoni, David Conrad
@@ -250,11 +241,8 @@ Codecs:
   wmv2.c                                Michael Niedermayer
   wnv1.c                                Kostya Shishkov
   xan.c                                 Mike Melanson
-  xbm*                                  Paul B Mahol
   xl.c                                  Kostya Shishkov
   xvmc.c                                Ivan Kalvachev
-  xwd*                                  Paul B Mahol
-  zerocodec.c                           Derek Buitenhuis
   zmbv*                                 Kostya Shishkov
 
 Hardware acceleration:
@@ -272,27 +260,22 @@ libavdevice
     libavdevice/avdevice.h
 
 
-  iec61883.c                            Georg Lippitsch
   libdc1394.c                           Roman Shaposhnik
   v4l2.c                                Luca Abeni
   vfwcap.c                              Ramiro Polla
-  dshow.c                               Roger Pack
+
 
 libavfilter
 ===========
 
-Generic parts:
+Video filters:
   graphdump.c                           Nicolas George
-
-Filters:
   af_amerge.c                           Nicolas George
   af_astreamsync.c                      Nicolas George
-  af_atempo.c                           Pavel Koshevoy
   af_pan.c                              Nicolas George
+  vsrc_mandelbrot.c                     Michael Niedermayer
   vf_yadif.c                            Michael Niedermayer
 
-Sources:
-  vsrc_mandelbrot.c                     Michael Niedermayer
 
 libavformat
 ===========
@@ -307,25 +290,17 @@ Generic parts:
 Muxers/Demuxers:
   4xm.c                                 Mike Melanson
   adtsenc.c                             Robert Swain
-  afc.c                                 Paul B Mahol
   aiff.c                                Baptiste Coudurier
   ape.c                                 Kostya Shishkov
   ass*                                  Aurelien Jacobs
-  astdec.c                              Paul B Mahol
-  astenc.c                              James Almer
   avi*                                  Michael Niedermayer
-  avr.c                                 Paul B Mahol
   bink.c                                Peter Ross
-  brstm.c                               Paul B Mahol
   caf*                                  Peter Ross
-  cdxl.c                                Paul B Mahol
   crc.c                                 Michael Niedermayer
   daud.c                                Reimar Doeffinger
-  dtshddec.c                            Paul B Mahol
   dv.c                                  Roman Shaposhnik
   dxa.c                                 Kostya Shishkov
   electronicarts.c                      Peter Ross
-  epafdec.c                             Paul B Mahol
   ffm*                                  Baptiste Coudurier
   flac*                                 Justin Ruggles
   flic.c                                Mike Melanson
@@ -336,22 +311,18 @@ Muxers/Demuxers:
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
   ipmovie.c                             Mike Melanson
-  img2*.c                               Michael Niedermayer
-  ircam*                                Paul B Mahol
+  img2.c                                Michael Niedermayer
   iss.c                                 Stefan Gehrer
-  jacosub*                              Clément Bœsch
   jvdec.c                               Peter Ross
   libmodplug.c                          Clément Bœsch
   libnut.c                              Oded Shimon
   lmlm4.c                               Ivo van Poorten
-  lvfdec.c                              Paul B Mahol
   lxfdec.c                              Tomas Härdin
   matroska.c                            Aurelien Jacobs
   matroskadec.c                         Aurelien Jacobs
   matroskaenc.c                         David Conrad
   metadata*                             Aurelien Jacobs
   microdvd*                             Aurelien Jacobs
-  mgsts.c                               Paul B Mahol
   mm.c                                  Peter Ross
   mov.c                                 Michael Niedermayer, Baptiste Coudurier
   movenc.c                              Michael Niedermayer, Baptiste Coudurier
@@ -363,7 +334,6 @@ Muxers/Demuxers:
   mtv.c                                 Reynaldo H. Verdejo Pinochet
   mxf*                                  Baptiste Coudurier
   mxfdec.c                              Tomas Härdin
-  nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
   nut.c                                 Michael Niedermayer
   nuv.c                                 Reimar Doeffinger
@@ -371,10 +341,8 @@ Muxers/Demuxers:
   oggenc.c                              Baptiste Coudurier
   oggparse*.c                           David Conrad
   oma.c                                 Maxim Poliakovski
-  paf.c                                 Paul B Mahol
   psxstr.c                              Mike Melanson
   pva.c                                 Ivo van Poorten
-  pvfdec.c                              Paul B Mahol
   r3d.c                                 Baptiste Coudurier
   raw.c                                 Michael Niedermayer
   rdt.c                                 Ronald S. Bultje
@@ -390,10 +358,8 @@ Muxers/Demuxers:
   segafilm.c                            Mike Melanson
   siff.c                                Kostya Shishkov
   smacker.c                             Kostya Shishkov
-  smjpeg*                               Paul B Mahol
   srtdec.c                              Aurelien Jacobs
   swf.c                                 Baptiste Coudurier
-  takdec.c                              Paul B Mahol
   tta.c                                 Alex Beregszaszi
   txd.c                                 Ivo van Poorten
   voc.c                                 Aurelien Jacobs
@@ -402,29 +368,13 @@ Muxers/Demuxers:
   westwood.c                            Mike Melanson
   wtv.c                                 Peter Ross
   wv.c                                  Kostya Shishkov
-  wvenc.c                               Paul B Mahol
 
 Protocols:
-  bluray.c                              Petri Hintukainen
   http.c                                Ronald S. Bultje
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
 
 
-libswresample
-=============
-
-Generic parts:
-  audioconvert.c                        Michael Niedermayer
-  dither.c                              Michael Niedermayer
-  rematrix*.c                           Michael Niedermayer
-  swresample*.c                         Michael Niedermayer
-
-Resamplers:
-  resample*.c                           Michael Niedermayer
-  soxr_resample.c                       Rob Sykes
-
-
 Operating systems / CPU architectures
 =====================================
 
@@ -445,11 +395,8 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
-1.2                                     Michael Niedermayer
-1.1                                     Michael Niedermayer
-1.0                                     Michael Niedermayer
+0.9                                     Michael Niedermayer
 
-If you want to maintain an older release, please contact us
 
 
 GnuPG Fingerprints of maintainers and contributors
@@ -461,7 +408,6 @@ Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
-Bœsch Clément                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
@@ -479,6 +425,5 @@ Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
-Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
-Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
-Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Stefano Sabatini              9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
+Tomas Härdin                  D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD

Makefile

@@ -15,13 +15,11 @@ PROGS-$(CONFIG_FFPLAY)   += ffplay
 PROGS-$(CONFIG_FFPROBE)  += ffprobe
 PROGS-$(CONFIG_FFSERVER) += ffserver
 
-PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS      := $(PROGS-yes:%=%$(EXESUF))
 INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-
-OBJS        = cmdutils.o $(EXEOBJS)
-OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
+OBJS        = $(PROGS-yes:%=%.o) cmdutils.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+HOSTPROGS  := $(TESTTOOLS:%=tests/%)
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
@@ -33,7 +31,6 @@ ALLMANPAGES = $(BASENAMES:%=%.1)
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
-FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_AVCODEC)  += avcodec
 FFLIBS-$(CONFIG_POSTPROC) += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
@@ -42,7 +39,6 @@ FFLIBS-$(CONFIG_SWSCALE)  += swscale
 FFLIBS := avutil
 
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
-EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
 
 SKIPHEADERS = cmdutils_common_opts.h
 
@@ -53,14 +49,14 @@ FF_DEP_LIBS  := $(DEP_LIBS)
 
 all: $(PROGS)
 
-$(PROGS): %$(EXESUF): %_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+$(PROGS): %$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+	$(CP) $< $@$(PROGSSUF)
+	$(STRIP) $@$(PROGSSUF)
 
-$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
+$(TOOLS): %$(EXESUF): %.o
+	$(LD) $(LDFLAGS) -o $@ $< $(ELIBS)
 
-tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
+tools/cws2fws$(EXESUF): ELIBS = -lz
 
 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -68,13 +64,9 @@ config.h: .config
 	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
 
-SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
-               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
-               ARMV5TE-OBJS ARMV6-OBJS VFP-OBJS NEON-OBJS                \
-               ALTIVEC-OBJS VIS-OBJS                                     \
-               MMX-OBJS YASM-OBJS                                        \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               OBJS HOSTOBJS TESTOBJS
+SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
+               ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
+               HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS ARMV6-OBJS TOOLS
 
 define RESET
 $(1) :=
@@ -91,19 +83,12 @@ endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
-define DOPROG
-OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
-$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
-$$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
-$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
-$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
--include $$(OBJS-$(1):.o=.d)
-endef
+ffplay.o: CFLAGS += $(SDL_CFLAGS)
+ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
+ffserver_g$(EXESUF): LDFLAGS += $(FFSERVERLDFLAGS)
 
-$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
-
-%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+%$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
 
 OBJDIRS += tools
 
@@ -137,10 +122,9 @@ install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 
-install-data: $(DATA_FILES) $(EXAMPLES_FILES)
-	$(Q)mkdir -p "$(DATADIR)/examples"
+install-data: $(DATA_FILES)
+	$(Q)mkdir -p "$(DATADIR)"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
-	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
 
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 
@@ -153,13 +137,14 @@ uninstall-data:
 clean::
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
+	$(RM) $(TOOLS)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) coverage.info
 	$(RM) -r coverage-html
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .version version.h libavutil/avconfig.h libavcodec/codec_names.h
+	$(RM) config.* .version version.h libavutil/avconfig.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
@@ -173,8 +158,6 @@ coverage-html: coverage.info
 	$(Q)genhtml -o $@ $<
 	$(Q)touch $@
 
-check: all alltools examples testprogs fate
-
 include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile
 
@@ -189,5 +172,5 @@ $(sort $(OBJDIRS)):
 # so this saves some time on slow systems.
 .SUFFIXES:
 
-.PHONY: all all-yes alltools check *clean config install*
+.PHONY: all all-yes alltools *clean config examples install*
 .PHONY: testprogs uninstall*

README

@@ -4,15 +4,9 @@ FFmpeg README
 1) Documentation
 ----------------
 
-* Read the documentation in the doc/ directory in git.
-  You can also view it online at http://ffmpeg.org/documentation.html
+* Read the documentation in the doc/ directory.
 
 2) Licensing
 ------------
 
 * See the LICENSE file.
-
-3) Build and Install
---------------------
-
-* See the INSTALL file.

RELEASE

@@ -1 +1 @@
-1.1.16
+0.10

VERSION

@@ -0,0 +1 @@
+0.10

arch.mak

@@ -1,16 +0,0 @@
-OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
-OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
-OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
-
-OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
-OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
-
-OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
-
-OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-OBJS-yes)
-
-OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
-OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.h

@@ -51,7 +51,6 @@ extern const int this_year;
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
-extern struct SwrContext *swr_opts;
 extern AVDictionary *format_opts, *codec_opts;
 
 /**
@@ -75,25 +74,23 @@ void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
  * Fallback for options that are not explicitly handled, these will be
  * parsed through AVOptions.
  */
-int opt_default(void *optctx, const char *opt, const char *arg);
+int opt_default(const char *opt, const char *arg);
 
 /**
  * Set the libav* libraries log level.
  */
-int opt_loglevel(void *optctx, const char *opt, const char *arg);
+int opt_loglevel(const char *opt, const char *arg);
 
 int opt_report(const char *opt);
 
-int opt_max_alloc(void *optctx, const char *opt, const char *arg);
+int opt_max_alloc(const char *opt, const char *arg);
 
-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
-
-int opt_codec_debug(void *optctx, const char *opt, const char *arg);
+int opt_codec_debug(const char *opt, const char *arg);
 
 /**
  * Limit the execution time.
  */
-int opt_timelimit(void *optctx, const char *opt, const char *arg);
+int opt_timelimit(const char *opt, const char *arg);
 
 /**
  * Parse a string and return its corresponding value as a double.
@@ -139,7 +136,7 @@ typedef struct SpecifierOpt {
     } u;
 } SpecifierOpt;
 
-typedef struct OptionDef {
+typedef struct {
     const char *name;
     int flags;
 #define HAS_ARG    0x0001
@@ -148,14 +145,14 @@ typedef struct OptionDef {
 #define OPT_STRING 0x0008
 #define OPT_VIDEO  0x0010
 #define OPT_AUDIO  0x0020
+#define OPT_GRAB   0x0040
 #define OPT_INT    0x0080
 #define OPT_FLOAT  0x0100
 #define OPT_SUBTITLE 0x0200
 #define OPT_INT64  0x0400
 #define OPT_EXIT   0x0800
 #define OPT_DATA   0x1000
-#define OPT_PERFILE  0x2000     /* the option is per-file (currently ffmpeg-only).
-                                   implied by OPT_OFFSET or OPT_SPEC */
+#define OPT_FUNC2  0x2000
 #define OPT_OFFSET 0x4000       /* option is specified as an offset in a passed optctx */
 #define OPT_SPEC   0x8000       /* option is to be stored in an array of SpecifierOpt.
                                    Implies OPT_OFFSET. Next element after the offset is
@@ -164,24 +161,16 @@ typedef struct OptionDef {
 #define OPT_DOUBLE 0x20000
      union {
         void *dst_ptr;
-        int (*func_arg)(void *, const char *, const char *);
+        int (*func_arg)(const char *, const char *);
+        int (*func2_arg)(void *, const char *, const char *);
         size_t off;
     } u;
     const char *help;
     const char *argname;
 } OptionDef;
 
-/**
- * Print help for all options matching specified flags.
- *
- * @param options a list of options
- * @param msg title of this group. Only printed if at least one option matches.
- * @param req_flags print only options which have all those flags set.
- * @param rej_flags don't print options which have any of those flags set.
- * @param alt_flags print only options that have at least one of those flags set
- */
-void show_help_options(const OptionDef *options, const char *msg, int req_flags,
-                       int rej_flags, int alt_flags);
+void show_help_options(const OptionDef *options, const char *msg, int mask,
+                       int value);
 
 /**
  * Show help for all options with given flags in class and all its
@@ -189,23 +178,10 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
  */
 void show_help_children(const AVClass *class, int flags);
 
-/**
- * Per-fftool specific help handler. Implemented in each
- * fftool, called by show_help().
- */
-void show_help_default(const char *opt, const char *arg);
-
-/**
- * Generic -h handler common to all fftools.
- */
-int show_help(void *optctx, const char *opt, const char *arg);
-
 /**
  * Parse the command line arguments.
  *
  * @param optctx an opaque options context
- * @param argc   number of command line arguments
- * @param argv   values of command line arguments
  * @param options Array with the definitions required to interpret every
  * option of the form: -option_name [argument]
  * @param parse_arg_function Name of the function called to process every
@@ -223,106 +199,11 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
 int parse_option(void *optctx, const char *opt, const char *arg,
                  const OptionDef *options);
 
-/**
- * An option extracted from the commandline.
- * Cannot use AVDictionary because of options like -map which can be
- * used multiple times.
- */
-typedef struct Option {
-    const OptionDef  *opt;
-    const char       *key;
-    const char       *val;
-} Option;
-
-typedef struct OptionGroupDef {
-    /**< group name */
-    const char *name;
-    /**
-     * Option to be used as group separator. Can be NULL for groups which
-     * are terminated by a non-option argument (e.g. ffmpeg output files)
-     */
-    const char *sep;
-} OptionGroupDef;
-
-typedef struct OptionGroup {
-    const OptionGroupDef *group_def;
-    const char *arg;
-
-    Option *opts;
-    int  nb_opts;
-
-    AVDictionary *codec_opts;
-    AVDictionary *format_opts;
-    struct SwsContext *sws_opts;
-    struct SwrContext *swr_opts;
-} OptionGroup;
-
-/**
- * A list of option groups that all have the same group type
- * (e.g. input files or output files)
- */
-typedef struct OptionGroupList {
-    const OptionGroupDef *group_def;
-
-    OptionGroup *groups;
-    int       nb_groups;
-} OptionGroupList;
-
-typedef struct OptionParseContext {
-    OptionGroup global_opts;
-
-    OptionGroupList *groups;
-    int           nb_groups;
-
-    /* parsing state */
-    OptionGroup cur_group;
-} OptionParseContext;
-
-/**
- * Parse an options group and write results into optctx.
- *
- * @param optctx an app-specific options context. NULL for global options group
- */
-int parse_optgroup(void *optctx, OptionGroup *g);
-
-/**
- * Split the commandline into an intermediate form convenient for further
- * processing.
- *
- * The commandline is assumed to be composed of options which either belong to a
- * group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
- * (everything else).
- *
- * A group (defined by an OptionGroupDef struct) is a sequence of options
- * terminated by either a group separator option (e.g. -i) or a parameter that
- * is not an option (doesn't start with -). A group without a separator option
- * must always be first in the supplied groups list.
- *
- * All options within the same group are stored in one OptionGroup struct in an
- * OptionGroupList, all groups with the same group definition are stored in one
- * OptionGroupList in OptionParseContext.groups. The order of group lists is the
- * same as the order of group definitions.
- */
-int split_commandline(OptionParseContext *octx, int argc, char *argv[],
-                      const OptionDef *options,
-                      const OptionGroupDef *groups, int nb_groups);
-
-/**
- * Free all allocated memory in an OptionParseContext.
- */
-void uninit_parse_context(OptionParseContext *octx);
-
 /**
  * Find the '-loglevel' option in the command line args and apply it.
  */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);
 
-/**
- * Return index of option opt in argv or 0 if not found.
- */
-int locate_option(int argc, char **argv, const OptionDef *options,
-                  const char *optname);
-
 /**
  * Check if the given stream matches a stream specifier.
  *
@@ -340,16 +221,12 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
  * Create a new options dictionary containing only the options from
  * opts which apply to the codec with ID codec_id.
  *
- * @param opts     dictionary to place options in
- * @param codec_id ID of the codec that should be filtered for
  * @param s Corresponding format context.
  * @param st A stream from s for which the options should be filtered.
- * @param codec The particular codec for which the options should be filtered.
- *              If null, the default one is looked up according to the codec id.
  * @return a pointer to the created dictionary
  */
-AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
-                                AVFormatContext *s, AVStream *st, AVCodec *codec);
+AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
+                                AVFormatContext *s, AVStream *st);
 
 /**
  * Setup AVCodecContext options for avformat_find_stream_info().
@@ -389,81 +266,62 @@ void show_banner(int argc, char **argv, const OptionDef *options);
  * libraries.
  * This option processing function does not utilize the arguments.
  */
-int show_version(void *optctx, const char *opt, const char *arg);
+int opt_version(const char *opt, const char *arg);
 
 /**
  * Print the license of the program to stdout. The license depends on
  * the license of the libraries compiled into the program.
  * This option processing function does not utilize the arguments.
  */
-int show_license(void *optctx, const char *opt, const char *arg);
+int opt_license(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the formats supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_formats(void *optctx, const char *opt, const char *arg);
+int opt_formats(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the codecs supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_codecs(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the decoders supported by the
- * program.
- */
-int show_decoders(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the encoders supported by the
- * program.
- */
-int show_encoders(void *optctx, const char *opt, const char *arg);
+int opt_codecs(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the filters supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_filters(void *optctx, const char *opt, const char *arg);
+int opt_filters(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the bit stream filters supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_bsfs(void *optctx, const char *opt, const char *arg);
+int opt_bsfs(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the protocols supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_protocols(void *optctx, const char *opt, const char *arg);
+int opt_protocols(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the pixel formats supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_pix_fmts(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the standard channel layouts supported by
- * the program.
- * This option processing function does not utilize the arguments.
- */
-int show_layouts(void *optctx, const char *opt, const char *arg);
+int opt_pix_fmts(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the sample formats supported by the
  * program.
  */
-int show_sample_fmts(void *optctx, const char *opt, const char *arg);
+int show_sample_fmts(const char *opt, const char *arg);
 
 /**
  * Return a positive value if a line read from standard input
@@ -475,7 +333,6 @@ int read_yesno(void);
  * Read the file with name filename, and put its content in a newly
  * allocated 0-terminated buffer.
  *
- * @param filename file to read from
  * @param bufptr location where pointer to buffer is returned
  * @param size   location where size of buffer is returned
  * @return 0 in case of success, a negative value corresponding to an
@@ -492,7 +349,7 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
  * at configuration time or in a "ffpresets" folder along the executable
  * on win32, in that order. If no such file is found and
  * codec_name is defined, then search for a file named
- * codec_name-preset_name.avpreset in the above-mentioned directories.
+ * codec_name-preset_name.ffpreset in the above-mentioned directories.
  *
  * @param filename buffer where the name of the found filename is written
  * @param filename_size size in bytes of the filename buffer
@@ -504,80 +361,20 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
 FILE *get_preset_file(char *filename, size_t filename_size,
                       const char *preset_name, int is_path, const char *codec_name);
 
+/**
+ * Do all the necessary cleanup and abort.
+ * This function is implemented in the avtools, not cmdutils.
+ */
+void exit_program(int ret);
+
 /**
  * Realloc array to hold new_size elements of elem_size.
- * Calls exit() on failure.
+ * Calls exit_program() on failure.
  *
- * @param array array to reallocate
  * @param elem_size size in bytes of each element
  * @param size new element count will be written here
- * @param new_size number of elements to place in reallocated array
  * @return reallocated array
  */
 void *grow_array(void *array, int elem_size, int *size, int new_size);
 
-#define GROW_ARRAY(array, nb_elems)\
-    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
-
-typedef struct FrameBuffer {
-    uint8_t *base[4];
-    uint8_t *data[4];
-    int  linesize[4];
-
-    int h, w;
-    enum AVPixelFormat pix_fmt;
-
-    int refcount;
-    struct FrameBuffer **pool;  ///< head of the buffer pool
-    struct FrameBuffer *next;
-} FrameBuffer;
-
-/**
- * Get a frame from the pool. This is intended to be used as a callback for
- * AVCodecContext.get_buffer.
- *
- * @param s codec context. s->opaque must be a pointer to the head of the
- *          buffer pool.
- * @param frame frame->opaque will be set to point to the FrameBuffer
- *              containing the frame data.
- */
-int codec_get_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVCodecContext.release_buffer along with
- * codec_get_buffer().
- */
-void codec_release_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVFilterBuffer.free.
- * @param fb buffer to free. fb->priv must be a pointer to the FrameBuffer
- *           containing the buffer data.
- */
-void filter_release_buffer(AVFilterBuffer *fb);
-
-/**
- * Free all the buffers in the pool. This must be called after all the
- * buffers have been released.
- */
-void free_buffer_pool(FrameBuffer **pool);
-
-#define GET_PIX_FMT_NAME(pix_fmt)\
-    const char *name = av_get_pix_fmt_name(pix_fmt);
-
-#define GET_SAMPLE_FMT_NAME(sample_fmt)\
-    const char *name = av_get_sample_fmt_name(sample_fmt)
-
-#define GET_SAMPLE_RATE_NAME(rate)\
-    char name[16];\
-    snprintf(name, sizeof(name), "%d", rate);
-
-#define GET_CH_LAYOUT_NAME(ch_layout)\
-    char name[16];\
-    snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
-
-#define GET_CH_LAYOUT_DESC(ch_layout)\
-    char name[128];\
-    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
-
 #endif /* CMDUTILS_H */

cmdutils_common_opts.h

@@ -1,21 +1,18 @@
-    { "L"          , OPT_EXIT, {.func_arg = show_license},      "show license" },
-    { "h"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "?"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
-    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
-    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
-    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
-    { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
-    { "bsfs"       , OPT_EXIT, {.func_arg = show_bsfs     },    "show available bit stream filters" },
-    { "protocols"  , OPT_EXIT, {.func_arg = show_protocols},    "show available protocols" },
-    { "filters"    , OPT_EXIT, {.func_arg = show_filters  },    "show available filters" },
-    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
-    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
+    { "L", OPT_EXIT, {(void*)opt_license}, "show license" },
+    { "h", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "?", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "help", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "-help", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "version", OPT_EXIT, {(void*)opt_version}, "show version" },
+    { "formats"  , OPT_EXIT, {(void*)opt_formats  }, "show available formats" },
+    { "codecs"   , OPT_EXIT, {(void*)opt_codecs   }, "show available codecs" },
+    { "bsfs"     , OPT_EXIT, {(void*)opt_bsfs     }, "show available bit stream filters" },
+    { "protocols", OPT_EXIT, {(void*)opt_protocols}, "show available protocols" },
+    { "filters",   OPT_EXIT, {(void*)opt_filters  }, "show available filters" },
+    { "pix_fmts" , OPT_EXIT, {(void*)opt_pix_fmts }, "show available pixel formats" },
     { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
-    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
-    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
-    { "report"     , 0,        {(void*)opt_report}, "generate a report" },
-    { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags"   , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },
+    { "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
+    { "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
+    { "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
+    { "report", 0, {(void*)opt_report}, "generate a report" },
+    { "max_alloc", HAS_ARG, {(void*)opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },

common.mak

@@ -10,9 +10,8 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
-SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
-
+BRIEF  = CC CXX AS YASM AR LD HOSTCC STRIP CP
+SILENT = DEPCC YASMDEP RM RANLIB
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
@@ -21,23 +20,21 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
 
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_PATH)/
 CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
 CFLAGS     += $(ECFLAGS)
-CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
-ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
-YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
-
-HOSTCCFLAGS = $(IFLAGS) $(HOSTCFLAGS)
-LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
+CCFLAGS     = $(CFLAGS)
+CXXFLAGS   := $(CFLAGS) $(CXXFLAGS)
+YASMFLAGS  += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
+HOSTCFLAGS += $(IFLAGS)
+LDFLAGS    := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)
 
 define COMPILE
-       $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
+       $($(1)DEP)
+       $($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
 endef
 
 COMPILE_C = $(call COMPILE,CC)
@@ -56,11 +53,8 @@ COMPILE_S = $(call COMPILE,AS)
 %.o: %.S
 	$(COMPILE_S)
 
-%.i: %.c
-	$(CC) $(CCFLAGS) $(CC_E) $<
-
-%.h.c:
-	$(Q)echo '#include "$*.h"' >$@
+%.ho: %.h
+	$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
 
 %.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
@@ -79,14 +73,13 @@ COMPILE_S = $(call COMPILE,AS)
 $(OBJS):
 endif
 
-include $(SRC_PATH)/arch.mak
+OBJS-$(HAVE_MMX) +=  $(MMX-OBJS-yes)
 
 OBJS      += $(OBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
-LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
-FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
+FFEXTRALIBS := $(FFLIBS:%=-l%$(BUILDSUF)) $(EXTRALIBS)
 
 EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
@@ -97,44 +90,31 @@ HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
 TOOLS     += $(TOOLS-yes)
 TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
-HEADERS   += $(HEADERS-yes)
 
 DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
 
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
-HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
-checkheaders: $(HOBJS)
-.SECONDARY:   $(HOBJS:.o=.c)
+checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
 
 alltools: $(TOOLS)
 
 $(HOSTOBJS): %.o: %.c
-	$(call COMPILE,HOSTCC)
+	$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
 
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
+	$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
-$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(TOOLOBJS): | tools
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
+CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
-LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
+LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
 
-define RULES
-clean::
-	$(RM) $(OBJS) $(OBJS:.o=.d)
-	$(RM) $(HOSTPROGS)
-	$(RM) $(TOOLS)
-endef
-
-$(eval $(RULES))
-
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))

compat/getopt.c

@@ -1,86 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * This file was copied from the following newsgroup posting:
- *
- * Newsgroups: mod.std.unix
- * Subject: public domain AT&T getopt source
- * Date: 3 Nov 85 19:34:15 GMT
- *
- * Here's something you've all been waiting for:  the AT&T public domain
- * source for getopt(3).  It is the code which was given out at the 1985
- * UNIFORUM conference in Dallas.  I obtained it by electronic mail
- * directly from AT&T.  The people there assure me that it is indeed
- * in the public domain.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-static int opterr = 1;
-static int optind = 1;
-static int optopt;
-static char *optarg;
-
-#undef fprintf
-
-static int getopt(int argc, char *argv[], char *opts)
-{
-    static int sp = 1;
-    int c;
-    char *cp;
-
-    if (sp == 1) {
-        if (optind >= argc ||
-            argv[optind][0] != '-' || argv[optind][1] == '\0')
-            return EOF;
-        else if (!strcmp(argv[optind], "--")) {
-            optind++;
-            return EOF;
-        }
-    }
-    optopt = c = argv[optind][sp];
-    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
-        fprintf(stderr, ": illegal option -- %c\n", c);
-        if (argv[optind][++sp] == '\0') {
-            optind++;
-            sp = 1;
-        }
-        return '?';
-    }
-    if (*++cp == ':') {
-        if (argv[optind][sp+1] != '\0')
-            optarg = &argv[optind++][sp+1];
-        else if(++optind >= argc) {
-            fprintf(stderr, ": option requires an argument -- %c\n", c);
-            sp = 1;
-            return '?';
-        } else
-            optarg = argv[optind++];
-        sp = 1;
-    } else {
-        if (argv[optind][++sp] == '\0') {
-            sp = 1;
-            optind++;
-        }
-        optarg = NULL;
-    }
-
-    return c;
-}

compat/msvcrt/snprintf.c

@@ -1,71 +0,0 @@
-/*
- * C99-compatible snprintf() and vsnprintf() implementations
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <stdio.h>
-#include <stdarg.h>
-#include <limits.h>
-#include <string.h>
-
-#include "compat/va_copy.h"
-#include "libavutil/error.h"
-
-#if defined(__MINGW32__)
-#define EOVERFLOW EFBIG
-#endif
-
-int avpriv_snprintf(char *s, size_t n, const char *fmt, ...)
-{
-    va_list ap;
-    int ret;
-
-    va_start(ap, fmt);
-    ret = avpriv_vsnprintf(s, n, fmt, ap);
-    va_end(ap);
-
-    return ret;
-}
-
-int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
-                     va_list ap)
-{
-    int ret;
-    va_list ap_copy;
-
-    if (n == 0)
-        return _vscprintf(fmt, ap);
-    else if (n > INT_MAX)
-        return AVERROR(EOVERFLOW);
-
-    /* we use n - 1 here because if the buffer is not big enough, the MS
-     * runtime libraries don't add a terminating zero at the end. MSDN
-     * recommends to provide _snprintf/_vsnprintf() a buffer size that
-     * is one less than the actual buffer, and zero it before calling
-     * _snprintf/_vsnprintf() to workaround this problem.
-     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
-    memset(s, 0, n);
-    va_copy(ap_copy, ap);
-    ret = _vsnprintf(s, n - 1, fmt, ap_copy);
-    va_end(ap_copy);
-    if (ret == -1)
-        ret = _vscprintf(fmt, ap);
-
-    return ret;
-}

compat/msvcrt/snprintf.h

@@ -1,38 +0,0 @@
-/*
- * C99-compatible snprintf() and vsnprintf() implementations
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef COMPAT_SNPRINTF_H
-#define COMPAT_SNPRINTF_H
-
-#include <stdarg.h>
-#include <stdio.h>
-
-int avpriv_snprintf(char *s, size_t n, const char *fmt, ...);
-int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
-
-#undef snprintf
-#undef _snprintf
-#undef vsnprintf
-#define snprintf avpriv_snprintf
-#define _snprintf avpriv_snprintf
-#define vsnprintf avpriv_vsnprintf
-
-#endif /* COMPAT_SNPRINTF_H */

compat/plan9/head

@@ -1,10 +0,0 @@
-#!/bin/sh
-
-n=10
-
-case "$1" in
-    -n)  n=$2;      shift 2 ;;
-    -n*) n=${1#-n}; shift   ;;
-esac
-
-exec sed ${n}q "$@"

compat/plan9/main.c

@@ -1,34 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-int plan9_main(int argc, char **argv);
-
-#undef main
-int main(int argc, char **argv)
-{
-    /* The setfcr() function in lib9 is broken, must use asm. */
-#ifdef __i386
-    short fcr;
-    __asm__ volatile ("fstcw        %0 \n"
-                      "or      $63, %0 \n"
-                      "fldcw        %0 \n"
-                      : "=m"(fcr));
-#endif
-
-    return plan9_main(argc, argv);
-}

compat/plan9/printf

@@ -1,2 +0,0 @@
-#!/bin/sh
-exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"

compat/strtod.c

@@ -1,94 +0,0 @@
-/*
- * C99-compatible strtod() implementation
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <ctype.h>
-#include <limits.h>
-#include <stdlib.h>
-
-#include "libavutil/avstring.h"
-#include "libavutil/mathematics.h"
-
-static char *check_nan_suffix(char *s)
-{
-    char *start = s;
-
-    if (*s++ != '(')
-        return start;
-
-    while ((*s >= 'a' && *s <= 'z') || (*s >= 'A' && *s <= 'Z') ||
-           (*s >= '0' && *s <= '9') ||  *s == '_')
-        s++;
-
-    return *s == ')' ? s + 1 : start;
-}
-
-#undef strtod
-double strtod(const char *, char **);
-
-double avpriv_strtod(const char *nptr, char **endptr)
-{
-    char *end;
-    double res;
-
-    /* Skip leading spaces */
-    while (isspace(*nptr))
-        nptr++;
-
-    if (!av_strncasecmp(nptr, "infinity", 8)) {
-        end = nptr + 8;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "inf", 3)) {
-        end = nptr + 3;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+infinity", 9)) {
-        end = nptr + 9;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+inf", 4)) {
-        end = nptr + 4;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "-infinity", 9)) {
-        end = nptr + 9;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "-inf", 4)) {
-        end = nptr + 4;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "nan", 3)) {
-        end = check_nan_suffix(nptr + 3);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "+nan", 4) ||
-               !av_strncasecmp(nptr, "-nan", 4)) {
-        end = check_nan_suffix(nptr + 4);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "0x", 2) ||
-               !av_strncasecmp(nptr, "-0x", 3) ||
-               !av_strncasecmp(nptr, "+0x", 3)) {
-        /* FIXME this doesn't handle exponents, non-integers (float/double)
-         * and numbers too large for long long */
-        res = strtoll(nptr, &end, 16);
-    } else {
-        res = strtod(nptr, &end);
-    }
-
-    if (endptr)
-        *endptr = end;
-
-    return res;
-}

compat/tms470/math.h

@@ -1,7 +0,0 @@
-#include_next <math.h>
-
-#undef INFINITY
-#undef NAN
-
-#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
-#define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))

doc/Makefile

@@ -1,45 +1,24 @@
-LIBRARIES-$(CONFIG_AVUTIL)     += libavutil
-LIBRARIES-$(CONFIG_SWSCALE)    += libswscale
-LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
-LIBRARIES-$(CONFIG_AVCODEC)    += libavcodec
-LIBRARIES-$(CONFIG_AVFORMAT)   += libavformat
-LIBRARIES-$(CONFIG_AVDEVICE)   += libavdevice
-LIBRARIES-$(CONFIG_AVFILTER)   += libavfilter
-
-COMPONENTS-yes = $(PROGS-yes)
-COMPONENTS-$(CONFIG_AVUTIL)     += ffmpeg-utils
-COMPONENTS-$(CONFIG_SWSCALE)    += ffmpeg-scaler
-COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
-COMPONENTS-$(CONFIG_AVCODEC)    += ffmpeg-codecs ffmpeg-bitstream-filters
-COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
-COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
-COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters
-
-MANPAGES    = $(COMPONENTS-yes:%=doc/%.1)    $(LIBRARIES-yes:%=doc/%.3)
-PODPAGES    = $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html)     \
+MANPAGES    = $(PROGS-yes:%=doc/%.1)
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
               doc/developer.html                                        \
               doc/faq.html                                              \
               doc/fate.html                                             \
               doc/general.html                                          \
               doc/git-howto.html                                        \
-              doc/nut.html                                              \
+              doc/libavfilter.html                                      \
               doc/platform.html                                         \
 
 TXTPAGES    = doc/fate.txt                                              \
 
 
-DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
-DOCS-$(CONFIG_PODPAGES)  += $(PODPAGES)
-DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
-DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
-DOCS = $(DOCS-yes)
+DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
+ifdef HAVE_MAKEINFO
+DOCS += $(TXTPAGES)
+endif
 
-all-$(CONFIG_DOC): doc
+all-$(CONFIG_DOC): documentation
 
-doc: documentation
-
-apidoc: doc/doxy/html
 documentation: $(DOCS)
 
 TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
@@ -49,55 +28,37 @@ doc/%.txt: doc/%.texi
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
 
-GENTEXI  = format codec
-GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
-
-$(GENTEXI): TAG = GENTEXI
-$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
-	$(M)doc/print_options $* > $@
-
 doc/%.html: TAG = HTML
-doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
+doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
 	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+	$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(GENTEXI)
+doc/%.pod: doc/%.texi
 	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
+	$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@
 
-doc/%.1 doc/%.3: TAG = MAN
-doc/%.1: doc/%.pod $(GENTEXI)
+doc/%.1: TAG = MAN
+doc/%.1: doc/%.pod
 	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
-doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
 
-$(DOCS) doc/doxy/html: | doc/
+$(DOCS): | doc
+OBJDIRS += doc
 
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
-
-install-man:
-
-ifdef CONFIG_MANPAGES
 install-progs-$(CONFIG_DOC): install-man
 
 install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
 	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
-endif
 
 uninstall: uninstall-man
 
 uninstall-man:
 	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
 
-clean:: docclean
-
-docclean:
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
-	$(RM) -r doc/doxy/html
+clean::
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
 
 -include $(wildcard $(DOCS:%=%.d))
 
-.PHONY: apidoc doc documentation
+.PHONY: documentation

doc/RELEASE_NOTES

@@ -1,11 +1,13 @@
 Release Notes
 =============
 
-* 1.1 "Fire Flower"  January, 2013
+* 0.10 "Freedom"  January, 2012
 
 
 General notes
 -------------
+This release is binary compatible with 0.8 and 0.9.
+
 See the Changelog file for a list of significant changes. Note, there
 are many more new features and bugfixes than whats listed there.
 
@@ -15,11 +17,33 @@ FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
 
-Of big interest to our Windows users, FFmpeg now supports building with the MSVC
-compiler. Since MSVC does not support C99 features used extensively by FFmpeg,
-this has been accomplished using a converter that turns C99 code to C89. See the
-platform-specific documentation for more detailed documentation on building
-FFmpeg with MSVC.
 
-The used output sample format for several audio decoders has changed, make
-sure you always check/use AVCodecContext.sample_fmt or AVFrame.format.
+API changes
+-----------
+
+A number of additional APIs have been introduced and some existing
+functions have been deprecated and are scheduled for removal in the next
+release. Significant API changes include:
+
+* new audio decoding API which decodes from an AVPacket to an AVFrame and
+is able to use AVCodecContext.get_buffer() in the similar way as video decoding.
+
+* new audio encoding API which encodes from an AVFrame to an AVPacket, thus
+allowing it to properly output timing information and side data.
+
+Please see the git history and the file doc/APIchanges for details.
+
+
+Other notable changes
+---------------------
+
+Libavcodec and libavformat built as shared libraries now hide non-public
+symbols. This will break applications using those symbols. Possible solutions
+are, in order of preference:
+1) Try finding a way of accomplishing the same with public API.
+2) If there is no corresponding public API, but you think there should be,
+post a request on the developer mailing list or IRC channel.
+3) Finally if your program needs access to FFmpeg / libavcodec / libavformat
+internals for some special reason then the best solution is to link statically.
+
+Please see the Changelog file and git history for a more detailed list of changes.

doc/authors.texi

@@ -1,11 +0,0 @@
-@chapter Authors
-
-The FFmpeg developers.
-
-For details about the authorship, see the Git history of the project
-(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
-@command{git log} in the FFmpeg source directory, or browsing the
-online repository at @url{http://source.ffmpeg.org}.
-
-Maintainers for the specific components are listed in the file
-@file{MAINTAINERS} in the source code tree.

doc/avtools-common-opts.texi

@@ -1,11 +1,10 @@
 All the numerical options, if not specified otherwise, accept in input
 a string representing a number, which may contain one of the
-SI unit prefixes, for example 'K', 'M', 'G'.
-If 'i' is appended after the prefix, binary prefixes are used,
-which are based on powers of 1024 instead of powers of 1000.
-The 'B' postfix multiplies the value by 8, and can be
-appended after a unit prefix or used alone. This allows using for
-example 'KB', 'MiB', 'G' and 'B' as number postfix.
+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
@@ -19,10 +18,10 @@ are used to precisely specify which stream(s) does a given option belong to.
 
 A stream specifier is a string generally appended to the option name and
 separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
-@code{a:1} stream specifier, which matches the second audio stream. Therefore it
+@code{a:1} stream specifer, which matches the second audio stream. Therefore it
 would select the ac3 codec for the second audio stream.
 
-A stream specifier can match several streams, the option is then applied to all
+A stream specifier can match several stream, the option is then applied to all
 of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
 streams.
 
@@ -42,10 +41,7 @@ streams of this type.
 @item p:@var{program_id}[:@var{stream_index}]
 If @var{stream_index} is given, then matches stream number @var{stream_index} in
 program with id @var{program_id}. Otherwise matches all streams in this program.
-@item #@var{stream_id}
-Matches the stream by format-specific ID.
 @end table
-
 @section Generic options
 
 These options are shared amongst the av* tools.
@@ -55,29 +51,8 @@ These options are shared amongst the av* tools.
 @item -L
 Show license.
 
-@item -h, -?, -help, --help [@var{arg}]
-Show help. An optional parameter may be specified to print help about a specific
-item.
-
-Possible values of @var{arg} are:
-@table @option
-@item decoder=@var{decoder_name}
-Print detailed information about the decoder named @var{decoder_name}. Use the
-@option{-decoders} option to get a list of all decoders.
-
-@item encoder=@var{encoder_name}
-Print detailed information about the encoder named @var{encoder_name}. Use the
-@option{-encoders} option to get a list of all encoders.
-
-@item demuxer=@var{demuxer_name}
-Print detailed information about the demuxer named @var{demuxer_name}. Use the
-@option{-formats} option to get a list of all demuxers and muxers.
-
-@item muxer=@var{muxer_name}
-Print detailed information about the muxer named @var{muxer_name}. Use the
-@option{-formats} option to get a list of all muxers and demuxers.
-
-@end table
+@item -h, -?, -help, --help
+Show help.
 
 @item -version
 Show version.
@@ -94,16 +69,23 @@ Encoding available
 @end table
 
 @item -codecs
-Show all codecs known to libavcodec.
+Show available codecs.
 
-Note that the term 'codec' is used throughout this documentation as a shortcut
-for what is more correctly called a media bitstream format.
-
-@item -decoders
-Show available decoders.
-
-@item -encoders
-Show all available encoders.
+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.
@@ -120,9 +102,6 @@ Show available pixel formats.
 @item -sample_fmts
 Show available sample formats.
 
-@item -layouts
-Show channel names and standard channel layouts.
-
 @item -loglevel @var{loglevel} | -v @var{loglevel}
 Set the logging level used by the library.
 @var{loglevel} is a number or a string containing one of the following values:
@@ -152,30 +131,8 @@ directory.
 This file can be useful for bug reports.
 It also implies @code{-loglevel verbose}.
 
-Setting the environment variable @code{FFREPORT} to any value has the
-same effect. If the value is a ':'-separated key=value sequence, these
-options will affect the report; options values must be escaped if they
-contain special characters or the options delimiter ':' (see the
-``Quoting and escaping'' section in the ffmpeg-utils manual). The
-following option is recognized:
-@table @option
-@item file
-set the file name to use for the report; @code{%p} is expanded to the name
-of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
-to a plain @code{%}
-@end table
-
-Errors in parsing the environment variable are not fatal, and will not
-appear in the report.
-
-@item -cpuflags flags (@emph{global})
-Allows setting and clearing cpu flags. This option is intended
-for testing. Do not use it unless you know what you're doing.
-@example
-ffmpeg -cpuflags -sse+mmx ...
-ffmpeg -cpuflags mmx ...
-ffmpeg -cpuflags 0 ...
-@end example
+Note: setting the environment variable @code{FFREPORT} to any value has the
+same effect.
 
 @end table
 

doc/bitstream_filters.texi

@@ -71,7 +71,7 @@ stream (carrying the AVI1 header ID and lacking a DHT segment) to
 produce fully qualified JPEG images.
 
 @example
-ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
+ffmpeg -i mjpeg-movie.avi -c:v copy -vbsf mjpeg2jpeg frame_%d.jpg
 exiftran -i -9 frame*.jpg
 ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 @end example

doc/decoders.texi

@@ -60,102 +60,4 @@ This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
 documented.
 
-@section libcelt
-
-libcelt decoder wrapper
-
-libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
-Requires the presence of the libcelt headers and library during configuration.
-You need to explicitly configure the build with @code{--enable-libcelt}.
-
-@section libgsm
-
-libgsm decoder wrapper
-
-libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
-the presence of the libgsm headers and library during configuration. You need
-to explicitly configure the build with @code{--enable-libgsm}.
-
-This decoder supports both the ordinary GSM and the Microsoft variant.
-
-@section libilbc
-
-libilbc decoder wrapper
-
-libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
-audio codec. Requires the presence of the libilbc headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libilbc}.
-
-@subsection Options
-
-The following option is supported by the libilbc wrapper.
-
-@table @option
-@item enhance
-
-Enable the enhancement of the decoded audio when set to 1. The default
-value is 0 (disabled).
-
-@end table
-
-@section libopencore-amrnb
-
-libopencore-amrnb decoder wrapper
-
-libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
-Narrowband audio codec. Using it requires the presence of the
-libopencore-amrnb headers and library during configuration. You need to
-explicitly configure the build with @code{--enable-libopencore-amrnb}.
-
-An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
-without this library.
-
-@section libopencore-amrwb
-
-libopencore-amrwb decoder wrapper.
-
-libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
-Wideband audio codec. Using it requires the presence of the
-libopencore-amrwb headers and library during configuration. You need to
-explicitly configure the build with @code{--enable-libopencore-amrwb}.
-
-An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
-without this library.
-
-@section libopus
-
-libopus decoder wrapper.
-
-libopus allows libavcodec to decode the Opus Interactive Audio Codec.
-Requires the presence of the libopus headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libopus}.
-
 @c man end AUDIO DECODERS
-
-@chapter Subtitles Decoders
-@c man begin SUBTILES DECODERS
-
-@section dvdsub
-
-This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
-also be found in VobSub file pairs and in some Matroska files.
-
-@subsection Options
-
-@table @option
-@item palette
-Specify the global palette used by the bitmaps. When stored in VobSub, the
-palette is normally specified in the index file; in Matroska, the palette is
-stored in the codec extra-data in the same format as in VobSub. In DVDs, the
-palette is stored in the IFO file, and therefore not available when reading
-from dumped VOB files.
-
-The format for this option is a string containing 16 24-bits hexadecimal
-numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
-ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
-7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
-@end table
-
-@c man end SUBTILES DECODERS

doc/default.css

@@ -1,149 +0,0 @@
-a {
-    color: #2D6198;
-}
-
-a:visited {
-    color: #884488;
-}
-
-#banner {
-    background-color: white;
-    position: relative;
-    text-align: center;
-}
-
-#banner img {
-    padding-bottom: 1px;
-    padding-top: 5px;
-}
-
-#body {
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-body {
-    background-color: #313131;
-    margin: 0;
-    text-align: justify;
-}
-
-.center {
-    margin-left: auto;
-    margin-right: auto;
-    text-align: center;
-}
-
-#container {
-    background-color: white;
-    color: #202020;
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-#footer {
-    text-align: center;
-}
-
-h1, h2, h3 {
-    padding-left: 0.4em;
-    border-radius: 4px;
-    padding-bottom: 0.2em;
-    padding-top: 0.2em;
-    border: 1px solid #6A996A;
-}
-
-h1 {
-    background-color: #7BB37B;
-    color: #151515;
-    font-size: 1.2em;
-    padding-bottom: 0.3em;
-    padding-top: 0.3em;
-}
-
-h2 {
-    color: #313131;
-    font-size: 0.9em;
-    background-color: #ABE3AB;
-}
-
-h3 {
-    color: #313131;
-    font-size: 0.8em;
-    margin-bottom: -8px;
-    background-color: #BBF3BB;
-}
-
-img {
-    border: 0;
-}
-
-#navbar {
-    background-color: #738073;
-    border-bottom: 1px solid #5C665C;
-    border-top: 1px solid #5C665C;
-    margin-top: 12px;
-    padding: 0.3em;
-    position: relative;
-    text-align: center;
-}
-
-#navbar a, #navbar_secondary a {
-    color: white;
-    padding: 0.3em;
-    text-decoration: none;
-}
-
-#navbar a:hover, #navbar_secondary a:hover {
-    background-color: #313131;
-    color: white;
-    text-decoration: none;
-}
-
-#navbar_secondary {
-    background-color: #738073;
-    border-bottom: 1px solid #5C665C;
-    border-left: 1px solid #5C665C;
-    border-right: 1px solid #5C665C;
-    padding: 0.3em;
-    position: relative;
-    text-align: center;
-}
-
-p {
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-pre {
-    margin-left: 3em;
-    margin-right: 3em;
-    padding: 0.3em;
-    border: 1px solid #bbb;
-    background-color: #f7f7f7;
-}
-
-dl dt {
-    font-weight: bold;
-}
-
-#proj_desc {
-    font-size: 1.2em;
-}
-
-#repos {
-    margin-left: 1em;
-    margin-right: 1em;
-    border-collapse: collapse;
-    border: solid 1px #6A996A;
-}
-
-#repos th {
-    background-color: #7BB37B;
-    border: solid 1px #6A996A;
-}
-
-#repos td {
-    padding: 0.2em;
-    border: solid 1px #6A996A;
-}

doc/demuxers.texi

@@ -23,31 +23,8 @@ The description of some of the currently available demuxers follows.
 Image file demuxer.
 
 This demuxer reads from a list of image files specified by a pattern.
-The syntax and meaning of the pattern is specified by the
-option @var{pattern_type}.
 
-The pattern may contain a suffix which is used to automatically
-determine the format of the images contained in the files.
-
-The size, the pixel format, and the format of each image must be the
-same for all the files in the sequence.
-
-This demuxer accepts the following options:
-@table @option
-@item framerate
-Set the framerate for the video stream. It defaults to 25.
-@item loop
-If set to 1, loop over the input. Default value is 0.
-@item pattern_type
-Select the pattern type used to interpret the provided filename.
-
-@var{pattern_type} accepts one of the following values.
-@table @option
-@item sequence
-Select a sequence pattern type, used to specify a sequence of files
-indexed by sequential numbers.
-
-A sequence pattern may contain the string "%d" or "%0@var{N}d", which
+The pattern may contain the string "%d" or "%0@var{N}d", which
 specifies the position of the characters representing a sequential
 number in each filename matched by the pattern. If the form
 "%d0@var{N}d" is used, the string representing the number in each
@@ -55,11 +32,13 @@ filename is 0-padded and @var{N} is the total number of 0-padded
 digits representing the number. The literal character '%' can be
 specified in the pattern with the string "%%".
 
-If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
+If the pattern contains "%d" or "%0@var{N}d", the first filename of
 the file list specified by the pattern must contain a number
-inclusively contained between @var{start_number} and
-@var{start_number}+@var{start_number_range}-1, and all the following
-numbers must be sequential.
+inclusively contained between 0 and 4, all the following numbers must
+be sequential. This limitation may be hopefully fixed.
+
+The pattern may contain a suffix which is used to automatically
+determine the format of the images contained in the files.
 
 For example the pattern "img-%03d.bmp" will match a sequence of
 filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@@ -67,6 +46,17 @@ filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
 sequence of filenames of the form @file{i%m%g-1.jpg},
 @file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
 
+The size, the pixel format, and the format of each image must be the
+same for all the files in the sequence.
+
+The following example shows how to use @command{ffmpeg} for creating a
+video from the images in the file sequence @file{img-001.jpeg},
+@file{img-002.jpeg}, ..., assuming an input frame rate of 10 frames per
+second:
+@example
+ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
+@end example
+
 Note that the pattern must not necessarily contain "%d" or
 "%0@var{N}d", for example to convert a single image file
 @file{img.jpeg} you can employ the command:
@@ -74,75 +64,6 @@ Note that the pattern must not necessarily contain "%d" or
 ffmpeg -i img.jpeg img.png
 @end example
 
-@item glob
-Select a glob wildcard pattern type.
-
-The pattern is interpreted like a @code{glob()} pattern. This is only
-selectable if libavformat was compiled with globbing support.
-
-@item glob_sequence @emph{(deprecated, will be removed)}
-Select a mixed glob wildcard/sequence pattern.
-
-If your version of libavformat was compiled with globbing support, and
-the provided pattern contains at least one glob meta character among
-@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
-interpreted like a @code{glob()} pattern, otherwise it is interpreted
-like a sequence pattern.
-
-All glob special characters @code{%*?[]@{@}} must be prefixed
-with "%". To escape a literal "%" you shall use "%%".
-
-For example the pattern @code{foo-%*.jpeg} will match all the
-filenames prefixed by "foo-" and terminating with ".jpeg", and
-@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
-"foo-", followed by a sequence of three characters, and terminating
-with ".jpeg".
-
-This pattern type is deprecated in favor of @var{glob} and
-@var{sequence}.
-@end table
-
-Default value is @var{glob_sequence}.
-@item pixel_format
-Set the pixel format of the images to read. If not specified the pixel
-format is guessed from the first image file in the sequence.
-@item start_number
-Set the index of the file matched by the image file pattern to start
-to read from. Default value is 0.
-@item start_number_range
-Set the index interval range to check when looking for the first image
-file in the sequence, starting from @var{start_number}. Default value
-is 5.
-@item video_size
-Set the video size of the images to read. If not specified the video
-size is guessed from the first image file in the sequence.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-Use @command{ffmpeg} for creating a video from the images in the file
-sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
-input frame rate of 10 frames per second:
-@example
-ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
-@end example
-
-@item
-As above, but start by reading from a file with index 100 in the sequence:
-@example
-ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
-@end example
-
-@item
-Read images matching the "*.png" glob pattern , that is all the files
-terminating with the ".png" suffix:
-@example
-ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
-@end example
-@end itemize
-
 @section applehttp
 
 Apple HTTP Live Streaming demuxer.
@@ -184,56 +105,4 @@ the script is directly played, the actual times will match the absolute
 timestamps up to the sound controller's clock accuracy, but if the user
 somehow pauses the playback or seeks, all times will be shifted accordingly.
 
-@section concat
-
-Virtual concatenation script demuxer.
-
-This demuxer reads a list of files and other directives from a text file and
-demuxes them one after the other, as if all their packet had been muxed
-together.
-
-The timestamps in the files are adjusted so that the first file starts at 0
-and each next file starts where the previous one finishes. Note that it is
-done globally and may cause gaps if all streams do not have exactly the same
-length.
-
-All files must have the same streams (same codecs, same time base, etc.).
-
-This script format can currently not be probed, it must be specified explicitly.
-
-@subsection Syntax
-
-The script is a text file in extended-ASCII, with one directive per line.
-Empty lines, leading spaces and lines starting with '#' are ignored. The
-following directive is recognized:
-
-@table @option
-
-@item @code{file @var{path}}
-Path to a file to read; special characters and spaces must be escaped with
-backslash or single quotes.
-
-@end table
-
-@section tedcaptions
-
-JSON captions used for @url{http://www.ted.com/, TED Talks}.
-
-TED does not provide links to the captions, but they can be guessed from the
-page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
-contains a bookmarklet to expose them.
-
-This demuxer accepts the following option:
-@table @option
-@item start_time
-Set the start time of the TED talk, in milliseconds. The default is 15000
-(15s). It is used to sync the captions with the downloadable videos, because
-they include a 15s intro.
-@end table
-
-Example: convert the captions to a format most players understand:
-@example
-ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
-@end example
-
 @c man end INPUT DEVICES

doc/developer.texi

@@ -14,13 +14,12 @@
 @section API
 @itemize @bullet
 @item libavcodec is the library containing the codecs (both encoding and
-decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
-it.
+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{doc/examples/muxing.c} to use it to generate audio or video
-streams.
+player. See @file{libavformat/output-example.c} to use it to generate
+audio or video streams.
 
 @end itemize
 
@@ -170,7 +169,7 @@ For exported names, each library has its own prefixes. Just check the existing
 code and name accordingly.
 @end itemize
 
-@subsection Miscellaneous conventions
+@subsection Miscellanous conventions
 @itemize @bullet
 @item
 fprintf and printf are forbidden in libavformat and libavcodec,
@@ -188,10 +187,8 @@ the following snippet into your @file{.vimrc}:
 set expandtab
 set shiftwidth=4
 set softtabstop=4
-set cindent
-set cinoptions=(0
-" Allow tabs in Makefiles.
-autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+" allow tabs in Makefiles
+autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
 " Trailing whitespace and tabs are forbidden, so highlight them.
 highlight ForbiddenWhitespace ctermbg=red guibg=red
 match ForbiddenWhitespace /\s\+$\|\t/
@@ -201,16 +198,10 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
 
 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 @example
-(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode . nil)
-               (show-trailing-whitespace . t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
+(setq c-default-style "k&r")
+(setq-default c-basic-offset 4)
+(setq-default indent-tabs-mode nil)
+(setq-default show-trailing-whitespace t)
 @end example
 
 @section Development Policy
@@ -317,8 +308,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
     (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). The third
-    component always starts at 100 to distinguish FFmpeg from Libav.
+    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
@@ -394,9 +384,7 @@ send a reminder by email. Your patch should eventually be dealt with.
 @item
     Did you register it in @file{allcodecs.c} or @file{allformats.c}?
 @item
-    Did you add the AVCodecID to @file{avcodec.h}?
-    When adding new codec IDs, also add an entry to the codec descriptor
-    list in @file{libavcodec/codec_desc.c}.
+    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?
@@ -430,7 +418,7 @@ send a reminder by email. Your patch should eventually be dealt with.
     Was the patch generated with git format-patch or send-email?
 @item
     Did you sign off your patch? (git commit -s)
-    See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+    See @url{http://kerneltrap.org/files/Jeremy/DCO.txt} for the meaning
     of sign off.
 @item
     Did you provide a clear git commit log message?
@@ -492,10 +480,6 @@ send a reminder by email. Your patch should eventually be dealt with.
     Consider to add a regression test for your code.
 @item
     If you added YASM code please check that things still work with --disable-yasm
-@item
-    Make sure you check the return values of function and return appropriate
-    error codes. Especially memory allocation functions like @code{av_malloc()}
-    are notoriously left unchecked, which is a serious problem.
 @end enumerate
 
 @section Patch review process
@@ -536,16 +520,4 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
 this case, the reference results of the regression tests shall be modified
 accordingly].
 
-@subsection Adding files to the fate-suite dataset
-
-When there is no muxer or encoder available to generate test media for a
-specific test then the media has to be inlcuded in the fate-suite.
-First please make sure that the sample file is as small as possible to test the
-respective decoder or demuxer sufficiently. Large files increase network
-bandwidth and disk space requirements.
-Once you have a working fate test and fate sample, provide in the commit
-message or introductionary message for the patch series that you post to
-the ffmpeg-devel mailing list, a direct link to download the sample media.
-
-
 @bye

doc/doxy-wrapper.sh

@@ -1,14 +0,0 @@
-#!/bin/sh
-
-SRC_PATH="${1}"
-DOXYFILE="${2}"
-
-shift 2
-
-doxygen - <<EOF
-@INCLUDE        = ${DOXYFILE}
-INPUT           = $@
-HTML_HEADER     = ${SRC_PATH}/doc/doxy/header.html
-HTML_FOOTER     = ${SRC_PATH}/doc/doxy/footer.html
-HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
-EOF

doc/doxy/footer.html

@@ -1,9 +1,10 @@
+ </div>
+
+ <div id="footer">
+   Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
+ </div>
 
- <footer class="footer pagination-right">
-   <span class="label label-info">
-    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
-   </span>
- </footer>
 </div>
+
 </body>
 </html>

doc/doxy/header.html

@@ -1,16 +1,14 @@
-<!DOCTYPE html>
-<html>
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
 <head>
-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
+<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
 <meta http-equiv="X-UA-Compatible" content="IE=9"/>
 <!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
 <!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
 <link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
-<!--Header replace -->
-
 </head>
 
-<div class="container">
+<div id="container">
 
-<!--Header replace -->
-<div class="menu">
+<div id="body">
+ <div>

doc/encoders.texi

@@ -25,95 +25,6 @@ enabled encoders.
 A description of some of the currently available audio encoders
 follows.
 
-@anchor{aacenc}
-@section aac
-
-Advanced Audio Coding (AAC) encoder.
-
-This encoder is an experimental FFmpeg-native AAC encoder. Currently only the
-low complexity (AAC-LC) profile is supported. To use this encoder, you must set
-@option{strict} option to @samp{experimental} or lower.
-
-As this encoder is experimental, unexpected behavior may exist from time to
-time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
-that it has a worse quality reported by some users.
-
-@c Comment this out until somebody writes the respective documentation.
-@c See also @ref{libfaac}, @ref{libaacplus}, and @ref{libfdk-aac-enc}.
-
-@subsection Options
-
-@table @option
-@item b
-Set bit rate in bits/s. Setting this automatically activates constant bit rate
-(CBR) mode.
-
-@item q
-Set quality for variable bit rate (VBR) mode. This option is valid only using
-the @command{ffmpeg} command-line tool. For library interface users, use
-@option{global_quality}.
-
-@item stereo_mode
-Set stereo encoding mode. Possible values:
-
-@table @samp
-@item auto
-Automatically selected by the encoder.
-
-@item ms_off
-Disable middle/side encoding. This is the default.
-
-@item ms_force
-Force middle/side encoding.
-@end table
-
-@item aac_coder
-Set AAC encoder coding method. Possible values:
-
-@table @samp
-@item 0
-FAAC-inspired method.
-
-This method is a simplified reimplementation of the method used in FAAC, which
-sets thresholds proportional to the band energies, and then decreases all the
-thresholds with quantizer steps to find the appropriate quantization with
-distortion below threshold band by band.
-
-The quality of this method is comparable to the two loop searching method
-descibed below, but somewhat a little better and slower.
-
-@item 1
-Average noise to mask ratio (ANMR) trellis-based solution.
-
-This has a theoretic best quality out of all the coding methods, but at the
-cost of the slowest speed.
-
-@item 2
-Two loop searching (TLS) method.
-
-This method first sets quantizers depending on band thresholds and then tries
-to find an optimal combination by adding or subtracting a specific value from
-all quantizers and adjusting some individual quantizer a little.
-
-This method produces similar quality with the FAAC method and is the default.
-
-@item 3
-Constant quantizer method.
-
-This method sets a constant quantizer for all bands. This is the fastest of all
-the methods, yet produces the worst quality.
-
-@end table
-
-@end table
-
-@subsection Tips and Tricks
-
-According to some reports
-(e.g. @url{http://d.hatena.ne.jp/kamedo2/20120729/1343545890}), setting the
-@option{cutoff} option to 15000 Hz greatly improves the quality of the output
-quality. As a result, we encourage you to do the same.
-
 @section ac3 and ac3_fixed
 
 AC-3 audio encoders.
@@ -501,279 +412,6 @@ Selected by Encoder (default)
 
 @end table
 
-@section libmp3lame
-
-LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper
-
-Requires the presence of the libmp3lame headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libmp3lame}.
-
-@subsection Options
-
-The following options are supported by the libmp3lame wrapper. The
-@command{lame}-equivalent of the options are listed in parentheses.
-
-@table @option
-@item b (@emph{-b})
-Set bitrate expressed in bits/s for CBR. LAME @code{bitrate} is
-expressed in kilobits/s.
-
-@item q (@emph{-V})
-Set constant quality setting for VBR. This option is valid only
-using the @command{ffmpeg} command-line tool. For library interface
-users, use @option{global_quality}.
-
-@item compression_level (@emph{-q})
-Set algorithm quality. Valid arguments are integers in the 0-9 range,
-with 0 meaning highest quality but slowest, and 9 meaning fastest
-while producing the worst quality.
-
-@item reservoir
-Enable use of bit reservoir when set to 1. Default value is 1. LAME
-has this enabled by default, but can be overriden by use
-@option{--nores} option.
-
-@end table
-
-@section libopencore-amrnb
-
-OpenCORE Adaptive Multi-Rate Narrowband encoder.
-
-Requires the presence of the libopencore-amrnb headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libopencore-amrnb --enable-version3}.
-
-This is a mono-only encoder. Officially it only supports 8000Hz sample rate,
-but you can override it by setting @option{strict} to @samp{unofficial} or
-lower.
-
-@subsection Options
-
-@table @option
-
-@item b
-Set bitrate in bits per second. Only the following bitrates are supported,
-otherwise libavcodec will round to the nearest valid bitrate.
-
-@table @option
-@item 4750
-@item 5150
-@item 5900
-@item 6700
-@item 7400
-@item 7950
-@item 10200
-@item 12200
-@end table
-
-@item dtx
-Allow discontinuous transmission (generate comfort noise) when set to 1. The
-default value is 0 (disabled).
-
-@end table
-
-@section libtwolame
-
-TwoLAME MP2 encoder wrapper
-
-Requires the presence of the libtwolame headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libtwolame}.
-
-@subsection Options
-
-The following options are supported by the libtwolame wrapper. The
-@command{twolame}-equivalent options follow the FFmpeg ones and are in
-parentheses.
-
-@table @option
-@item b (@emph{-b})
-Set bitrate expressed in bits/s for CBR. @command{twolame} @option{b}
-option is expressed in kilobits/s. Default value is 128k.
-
-@item q (@emph{-V})
-Set quality for experimental VBR support. Maximum value range is
-from -50 to 50, useful range is from -10 to 10. The higher the
-value, the better the quality. This option is valid only using the
-@command{ffmpeg} command-line tool. For library interface users,
-use @option{global_quality}.
-
-@item mode (@emph{--mode})
-Set the mode of the resulting audio. Possible values:
-
-@table @samp
-@item auto
-Choose mode automatically based on the input. This is the default.
-@item stereo
-Stereo
-@item joint_stereo
-Joint stereo
-@item dual_channel
-Dual channel
-@item mono
-Mono
-@end table
-
-@item psymodel (@emph{--psyc-mode})
-Set psychoacoustic model to use in encoding. The argument must be
-an integer between -1 and 4, inclusive. The higher the value, the
-better the quality. The default value is 3.
-
-@item energy_levels (@emph{--energy})
-Enable energy levels extensions when set to 1. The default value is
-0 (disabled).
-
-@item error_protection (@emph{--protect})
-Enable CRC error protection when set to 1. The default value is 0
-(disabled).
-
-@item copyright (@emph{--copyright})
-Set MPEG audio copyright flag when set to 1. The default value is 0
-(disabled).
-
-@item original (@emph{--original})
-Set MPEG audio original flag when set to 1. The default value is 0
-(disabled).
-
-@end table
-
-@anchor{libvo-aacenc}
-@section libvo-aacenc
-
-VisualOn AAC encoder
-
-Requires the presence of the libvo-aacenc headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libvo-aacenc --enable-version3}.
-
-This encoder is considered to be worse than the
-@ref{aacenc,,native experimental FFmpeg AAC encoder}, according to
-multiple sources.
-
-@subsection Options
-
-The VisualOn AAC encoder only support encoding AAC-LC and up to 2
-channels. It is also CBR-only.
-
-@table @option
-
-@item b
-Set bit rate in bits/s.
-
-@end table
-
-@section libvo-amrwbenc
-
-VisualOn Adaptive Multi-Rate Wideband encoder
-
-Requires the presence of the libvo-amrwbenc headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libvo-amrwbenc --enable-version3}.
-
-This is a mono-only encoder. Officially it only supports 16000Hz sample
-rate, but you can override it by setting @option{strict} to
-@samp{unofficial} or lower.
-
-@subsection Options
-
-@table @option
-
-@item b
-Set bitrate in bits/s. Only the following bitrates are supported, otherwise
-libavcodec will round to the nearest valid bitrate.
-
-@table @samp
-@item 6600
-@item 8850
-@item 12650
-@item 14250
-@item 15850
-@item 18250
-@item 19850
-@item 23050
-@item 23850
-@end table
-
-@item dtx
-Allow discontinuous transmission (generate comfort noise) when set to 1. The
-default value is 0 (disabled).
-
-@end table
-
-@section libopus
-
-libopus Opus Interactive Audio Codec encoder wrapper.
-
-Requires the presence of the libopus headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libopus}.
-
-@subsection Option Mapping
-
-Most libopus options are modeled after the @command{opusenc} utility from
-opus-tools. The following is an option mapping chart describing options
-supported by the libopus wrapper, and their @command{opusenc}-equivalent
-in parentheses.
-
-@table @option
-
-@item b (@emph{bitrate})
-Set the bit rate in bits/s.  FFmpeg's @option{b} option is
-expressed in bits/s, while @command{opusenc}'s @option{bitrate} in
-kilobits/s.
-
-@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr})
-Set VBR mode. The FFmpeg @option{vbr} option has the following
-valid arguments, with the their @command{opusenc} equivalent options
-in parentheses:
-
-@table @samp
-@item off (@emph{hard-cbr})
-Use constant bit rate encoding.
-
-@item on (@emph{vbr})
-Use variable bit rate encoding (the default).
-
-@item constrained (@emph{cvbr})
-Use constrained variable bit rate encoding.
-@end table
-
-@item compression_level (@emph{comp})
-Set encoding algorithm complexity. Valid options are integers in
-the 0-10 range. 0 gives the fastest encodes but lower quality, while 10
-gives the highest quality but slowest encoding. The default is 10.
-
-@item frame_duration (@emph{framesize})
-Set maximum frame size, or duration of a frame in milliseconds. The
-argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
-frame sizes achieve lower latency but less quality at a given bitrate.
-Sizes greater than 20ms are only interesting at fairly low bitrates.
-The default is 20ms.
-
-@item packet_loss (@emph{expect-loss})
-Set expected packet loss percentage. The default is 0.
-
-@item application (N.A.)
-Set intended application type. Valid options are listed below:
-
-@table @samp
-@item voip
-Favor improved speech intelligibility.
-@item audio
-Favor faithfulness to the input (the default).
-@item lowdelay
-Restrict to only the lowest delay modes.
-@end table
-
-@item cutoff (N.A.)
-Set cutoff bandwidth in Hz. The argument must be exactly one of the
-following: 4000, 6000, 8000, 12000, or 20000, corresponding to
-narrowband, mediumband, wideband, super wideband, and fullband
-respectively. The default is 0 (cutoff disabled).
-
-@end table
-
 @c man end AUDIO ENCODERS
 
 @chapter Video Encoders
@@ -782,46 +420,6 @@ respectively. The default is 0 (cutoff disabled).
 A description of some of the currently available video encoders
 follows.
 
-@section libtheora
-
-Theora format supported through libtheora.
-
-Requires the presence of the libtheora headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libtheora}.
-
-@subsection Options
-
-The following global options are mapped to internal libtheora options
-which affect the quality and the bitrate of the encoded stream.
-
-@table @option
-@item b
-Set the video bitrate, only works if the @code{qscale} flag in
-@option{flags} is not enabled.
-
-@item flags
-Used to enable constant quality mode encoding through the
-@option{qscale} flag, and to enable the @code{pass1} and @code{pass2}
-modes.
-
-@item g
-Set the GOP size.
-
-@item global_quality
-Set the global quality in lambda units, only works if the
-@code{qscale} flag in @option{flags} is enabled. The value is clipped
-in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3
-to get a value in the native libtheora range [0-63]. A higher value
-corresponds to a higher quality.
-
-For example, to set maximum constant quality encoding with
-@command{ffmpeg}:
-@example
-ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg
-@end example
-@end table
-
 @section libvpx
 
 VP8 format supported through libvpx.
@@ -913,12 +511,6 @@ rc_2pass_vbr_minsection_pct
 @item slices
 @code{VP8E_SET_TOKEN_PARTITIONS}
 
-@item max-intra-rate
-@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
-
-@item force_key_frames
-@code{VPX_EFLAG_FORCE_KF}
-
 @item Alternate reference frame related
 @table @option
 @item vp8flags altref
@@ -979,12 +571,10 @@ Specify Weighted prediction for P-frames.
 Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).
 
 @item x264opts @var{options}
-Allow to set any x264 option, see @code{x264 --fullhelp} for a list.
+Allow to set any x264 option, see x264 --fullhelp for a list.
 
 @var{options} is a list of @var{key}=@var{value} couples separated by
-":". In @var{filter} and @var{psy-rd} options that use ":" as a separator
-themselves, use "," instead. They accept it as well since long ago but this
-is kept undocumented for some reason.
+":".
 @end table
 
 For example to specify libx264 encoding options with @command{ffmpeg}:
@@ -995,117 +585,4 @@ ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
 For more information about libx264 and the supported options see:
 @url{http://www.videolan.org/developers/x264.html}
 
-@section libxvid
-
-Xvid MPEG-4 Part 2 encoder wrapper.
-
-This encoder requires the presence of the libxvidcore headers and library
-during configuration. You need to explicitly configure the build with
-@code{--enable-libxvid --enable-gpl}.
-
-The native @code{mpeg4} encoder supports the MPEG-4 Part 2 format, so
-users can encode to this format without this library.
-
-@subsection Options
-
-The following options are supported by the libxvid wrapper. Some of
-the following options are listed but are not documented, and
-correspond to shared codec options. See @ref{codec-options,,the Codec
-Options chapter} for their documentation. The other shared options
-which are not listed have no effect for the libxvid encoder.
-
-@table @option
-@item b
-
-@item g
-
-@item qmin
-
-@item qmax
-
-@item mpeg_quant
-
-@item threads
-
-@item bf
-
-@item b_qfactor
-
-@item b_qoffset
-
-@item flags
-Set specific encoding flags. Possible values:
-
-@table @samp
-
-@item mv4
-Use four motion vector by macroblock.
-
-@item aic
-Enable high quality AC prediction.
-
-@item gray
-Only encode grayscale.
-
-@item gmc
-Enable the use of global motion compensation (GMC).
-
-@item qpel
-Enable quarter-pixel motion compensation.
-
-@item cgop
-Enable closed GOP.
-
-@item global_header
-Place global headers in extradata instead of every keyframe.
-
-@end table
-
-@item trellis
-
-@item me_method
-Set motion estimation method. Possible values in decreasing order of
-speed and increasing order of quality:
-
-@table @samp
-@item zero
-Use no motion estimation (default).
-
-@item phods
-@item x1
-@item log
-Enable advanced diamond zonal search for 16x16 blocks and half-pixel
-refinement for 16x16 blocks. @samp{x1} and @samp{log} are aliases for
-@samp{phods}.
-
-@item epzs
-Enable all of the things described above, plus advanced diamond zonal
-search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion
-estimation on chroma planes.
-
-@item full
-Enable all of the things described above, plus extended 16x16 and 8x8
-blocks search.
-@end table
-
-@item mbd
-Set macroblock decision algorithm. Possible values in the increasing
-order of quality:
-
-@table @samp
-@item simple
-Use macroblock comparing function algorithm (default).
-
-@item bits
-Enable rate distortion-based half pixel and quarter pixel refinement for
-16x16 blocks.
-
-@item rd
-Enable all of the things described above, plus rate distortion-based
-half pixel and quarter pixel refinement for 8x8 blocks, and rate
-distortion-based search using square pattern.
-@end table
-
-@end table
-
 @c man end VIDEO ENCODERS

doc/eval.texi

@@ -21,78 +21,30 @@ The following unary operators are available: @code{+}, @code{-}.
 The following functions are available:
 @table @option
 @item sinh(x)
-Compute hyperbolic sine of @var{x}.
-
 @item cosh(x)
-Compute hyperbolic cosine of @var{x}.
-
 @item tanh(x)
-Compute hyperbolic tangent of @var{x}.
-
 @item sin(x)
-Compute sine of @var{x}.
-
 @item cos(x)
-Compute cosine of @var{x}.
-
 @item tan(x)
-Compute tangent of @var{x}.
-
 @item atan(x)
-Compute arctangent of @var{x}.
-
 @item asin(x)
-Compute arcsine of @var{x}.
-
 @item acos(x)
-Compute arccosine of @var{x}.
-
 @item exp(x)
-Compute exponential of @var{x} (with base @code{e}, the Euler's number).
-
 @item log(x)
-Compute natural logarithm of @var{x}.
-
 @item abs(x)
-Compute absolute value of @var{x}.
-
 @item squish(x)
-Compute expression @code{1/(1 + exp(4*x))}.
-
 @item gauss(x)
-Compute Gauss function of @var{x}, corresponding to
-@code{exp(-x*x/2) / sqrt(2*PI)}.
-
-@item isinf(x)
-Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
-
 @item isnan(x)
 Return 1.0 if @var{x} is NAN, 0.0 otherwise.
 
 @item mod(x, y)
-Compute the remainder of division of @var{x} by @var{y}.
-
 @item max(x, y)
-Return the maximum between @var{x} and @var{y}.
-
 @item min(x, y)
-Return the maximum between @var{x} and @var{y}.
-
 @item eq(x, y)
-Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
-
 @item gte(x, y)
-Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
-
 @item gt(x, y)
-Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
-
 @item lte(x, y)
-Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
-
 @item lt(x, y)
-Return 1 if @var{x} is lesser than @var{y}, 0 otherwise.
-
 @item st(var, expr)
 Allow to store the value of the expression @var{expr} in an internal
 variable. @var{var} specifies the number of the variable where to
@@ -154,18 +106,6 @@ the evaluation of @var{y}, return 0 otherwise.
 @item ifnot(x, y)
 Evaluate @var{x}, and if the result is zero return the result of the
 evaluation of @var{y}, return 0 otherwise.
-
-@item taylor(expr, x) taylor(expr, x, id)
-Evaluate a taylor series at x.
-expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
-then 0 is assumed.
-note, when you have the derivatives at y instead of 0
-taylor(expr, x-y) can be used
-When the series does not converge the results are undefined.
-
-@item root(expr, max)
-Finds x where f(x)=0 in the interval 0..max.
-f() must be continuous or the result is undefined.
 @end table
 
 The following constants are available:

doc/examples/Makefile

@@ -1,37 +1,21 @@
-# use pkg-config for getting CFLAGS and LDLIBS
-FFMPEG_LIBS=    libavdevice                        \
-                libavformat                        \
-                libavfilter                        \
-                libavcodec                         \
-                libswresample                      \
-                libswscale                         \
-                libavutil                          \
+# use pkg-config for getting CFLAGS abd LDFLAGS
+FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
+CFLAGS+=$(shell pkg-config  --cflags $(FFMPEG_LIBS))
+LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
 
-CFLAGS += -Wall -O2 -g
-CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
-LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
-
-EXAMPLES=       decoding_encoding                  \
-                demuxing                           \
-                filtering_video                    \
-                filtering_audio                    \
-                metadata                           \
-                muxing                             \
-                resampling_audio                   \
-                scaling_video                      \
+EXAMPLES=decoding_encoding filtering metadata muxing
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
-# the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
-muxing:            LDLIBS += -lm
+%: %.o
+	$(CC) $< $(LDFLAGS) -o $@
 
-.phony: all clean-test clean
+%.o: %.c
+	$(CC) $< $(CFLAGS) -c -o $@
+
+.phony: all clean
 
 all: $(OBJS) $(EXAMPLES)
 
-clean-test:
-	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
-
-clean: clean-test
-	$(RM) $(EXAMPLES) $(OBJS)
+clean:
+	rm -rf $(EXAMPLES) $(OBJS)

doc/examples/README

@@ -1,18 +0,0 @@
-FFmpeg examples README
-----------------------
-
-Both following use cases rely on pkg-config and make, thus make sure
-that you have them installed and working on your system.
-
-
-1) Build the installed examples in a generic read/write user directory
-
-Copy to a read/write user directory and just use "make", it will link
-to the libraries on your system, assuming the PKG_CONFIG_PATH is
-correctly configured.
-
-2) Build the examples in-tree
-
-Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then you can go into the
-doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

doc/examples/decoding_encoding.c

@@ -27,76 +27,18 @@
  * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
  * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
  * format handling
- * @example doc/examples/decoding_encoding.c
  */
 
-#include <math.h>
-
-#include <libavutil/opt.h>
-#include <libavcodec/avcodec.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/common.h>
-#include <libavutil/imgutils.h>
-#include <libavutil/mathematics.h>
-#include <libavutil/samplefmt.h>
+#include "libavutil/imgutils.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avcodec.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/samplefmt.h"
 
 #define INBUF_SIZE 4096
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 
-/* check that a given sample format is supported by the encoder */
-static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
-{
-    const enum AVSampleFormat *p = codec->sample_fmts;
-
-    while (*p != AV_SAMPLE_FMT_NONE) {
-        if (*p == sample_fmt)
-            return 1;
-        p++;
-    }
-    return 0;
-}
-
-/* just pick the highest supported samplerate */
-static int select_sample_rate(AVCodec *codec)
-{
-    const int *p;
-    int best_samplerate = 0;
-
-    if (!codec->supported_samplerates)
-        return 44100;
-
-    p = codec->supported_samplerates;
-    while (*p) {
-        best_samplerate = FFMAX(*p, best_samplerate);
-        p++;
-    }
-    return best_samplerate;
-}
-
-/* select layout with the highest channel count */
-static int select_channel_layout(AVCodec *codec)
-{
-    const uint64_t *p;
-    uint64_t best_ch_layout = 0;
-    int best_nb_channells   = 0;
-
-    if (!codec->channel_layouts)
-        return AV_CH_LAYOUT_STEREO;
-
-    p = codec->channel_layouts;
-    while (*p) {
-        int nb_channels = av_get_channel_layout_nb_channels(*p);
-
-        if (nb_channels > best_nb_channells) {
-            best_ch_layout    = *p;
-            best_nb_channells = nb_channels;
-        }
-        p++;
-    }
-    return best_ch_layout;
-}
-
 /*
  * Audio encoding example
  */
@@ -104,83 +46,44 @@ static void audio_encode_example(const char *filename)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    AVFrame *frame;
-    AVPacket pkt;
-    int i, j, k, ret, got_output;
-    int buffer_size;
+    int frame_size, i, j, out_size, outbuf_size;
     FILE *f;
-    uint16_t *samples;
+    short *samples;
     float t, tincr;
+    uint8_t *outbuf;
 
-    printf("Encode audio file %s\n", filename);
+    printf("Audio encoding\n");
 
     /* find the MP2 encoder */
-    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
+    codec = avcodec_find_encoder(CODEC_ID_MP2);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
 
     /* put sample parameters */
     c->bit_rate = 64000;
-
-    /* check that the encoder supports s16 pcm input */
+    c->sample_rate = 44100;
+    c->channels = 2;
     c->sample_fmt = AV_SAMPLE_FMT_S16;
-    if (!check_sample_fmt(codec, c->sample_fmt)) {
-        fprintf(stderr, "Encoder does not support sample format %s",
-                av_get_sample_fmt_name(c->sample_fmt));
-        exit(1);
-    }
-
-    /* select other audio parameters supported by the encoder */
-    c->sample_rate    = select_sample_rate(codec);
-    c->channel_layout = select_channel_layout(codec);
-    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
+    /* the codec gives us the frame size, in samples */
+    frame_size = c->frame_size;
+    samples = malloc(frame_size * 2 * c->channels);
+    outbuf_size = 10000;
+    outbuf = malloc(outbuf_size);
+
     f = fopen(filename, "wb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    /* frame containing input raw audio */
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate audio frame\n");
-        exit(1);
-    }
-
-    frame->nb_samples     = c->frame_size;
-    frame->format         = c->sample_fmt;
-    frame->channel_layout = c->channel_layout;
-
-    /* the codec gives us the frame size, in samples,
-     * we calculate the size of the samples buffer in bytes */
-    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
-                                             c->sample_fmt, 0);
-    samples = av_malloc(buffer_size);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
-                buffer_size);
-        exit(1);
-    }
-    /* setup the data pointers in the AVFrame */
-    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                                   (const uint8_t*)samples, buffer_size, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not setup audio frame\n");
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
 
@@ -188,46 +91,19 @@ static void audio_encode_example(const char *filename)
     t = 0;
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
     for(i=0;i<200;i++) {
-        av_init_packet(&pkt);
-        pkt.data = NULL; // packet data will be allocated by the encoder
-        pkt.size = 0;
-
-        for (j = 0; j < c->frame_size; j++) {
+        for(j=0;j<frame_size;j++) {
             samples[2*j] = (int)(sin(t) * 10000);
-
-            for (k = 1; k < c->channels; k++)
-                samples[2*j + k] = samples[2*j];
+            samples[2*j+1] = samples[2*j];
             t += tincr;
         }
         /* encode the samples */
-        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding audio frame\n");
-            exit(1);
-        }
-        if (got_output) {
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
-    }
-
-    /* get the delayed frames */
-    for (got_output = 1; got_output; i++) {
-        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
+        out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
+        fwrite(outbuf, 1, out_size, f);
     }
     fclose(f);
+    free(outbuf);
+    free(samples);
 
-    av_freep(&samples);
-    avcodec_free_frame(&frame);
     avcodec_close(c);
     av_free(c);
 }
@@ -247,30 +123,26 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     av_init_packet(&avpkt);
 
-    printf("Decode audio file %s to %s\n", filename, outfilename);
+    printf("Audio decoding\n");
 
     /* find the mpeg audio decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
+    codec = avcodec_find_decoder(CODEC_ID_MP2);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     f = fopen(filename, "rb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
     outfile = fopen(outfilename, "wb");
@@ -288,7 +160,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
         if (!decoded_frame) {
             if (!(decoded_frame = avcodec_alloc_frame())) {
-                fprintf(stderr, "Could not allocate audio frame\n");
+                fprintf(stderr, "out of memory\n");
                 exit(1);
             }
         } else
@@ -329,7 +201,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&decoded_frame);
+    av_free(decoded_frame);
 }
 
 /*
@@ -339,26 +211,22 @@ static void video_encode_example(const char *filename, int codec_id)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    int i, ret, x, y, got_output;
+    int i, out_size, size, x, y, outbuf_size;
     FILE *f;
-    AVFrame *frame;
-    AVPacket pkt;
-    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
+    AVFrame *picture;
+    uint8_t *outbuf;
 
-    printf("Encode video file %s\n", filename);
+    printf("Video encoding\n");
 
     /* find the mpeg1 video encoder */
     codec = avcodec_find_encoder(codec_id);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
+    picture= avcodec_alloc_frame();
 
     /* put sample parameters */
     c->bit_rate = 400000;
@@ -369,105 +237,79 @@ static void video_encode_example(const char *filename, int codec_id)
     c->time_base= (AVRational){1,25};
     c->gop_size = 10; /* emit one intra frame every ten frames */
     c->max_b_frames=1;
-    c->pix_fmt = AV_PIX_FMT_YUV420P;
+    c->pix_fmt = PIX_FMT_YUV420P;
 
-    if(codec_id == AV_CODEC_ID_H264)
+    if(codec_id == CODEC_ID_H264)
         av_opt_set(c->priv_data, "preset", "slow", 0);
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     f = fopen(filename, "wb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
 
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-    frame->format = c->pix_fmt;
-    frame->width  = c->width;
-    frame->height = c->height;
+    /* alloc image and output buffer */
+    outbuf_size = 100000;
+    outbuf = malloc(outbuf_size);
 
     /* the image can be allocated by any means and av_image_alloc() is
      * just the most convenient way if av_malloc() is to be used */
-    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
-                         c->pix_fmt, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate raw picture buffer\n");
-        exit(1);
-    }
+    av_image_alloc(picture->data, picture->linesize,
+                   c->width, c->height, c->pix_fmt, 1);
 
     /* encode 1 second of video */
     for(i=0;i<25;i++) {
-        av_init_packet(&pkt);
-        pkt.data = NULL;    // packet data will be allocated by the encoder
-        pkt.size = 0;
-
         fflush(stdout);
         /* prepare a dummy image */
         /* Y */
         for(y=0;y<c->height;y++) {
             for(x=0;x<c->width;x++) {
-                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
+                picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3;
             }
         }
 
         /* Cb and Cr */
         for(y=0;y<c->height/2;y++) {
             for(x=0;x<c->width/2;x++) {
-                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
-                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
+                picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
+                picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
             }
         }
 
-        frame->pts = i;
-
         /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
+        out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
+        printf("encoding frame %3d (size=%5d)\n", i, out_size);
+        fwrite(outbuf, 1, out_size, f);
     }
 
     /* get the delayed frames */
-    for (got_output = 1; got_output; i++) {
+    for(; out_size; i++) {
         fflush(stdout);
 
-        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
+        out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
+        printf("write frame %3d (size=%5d)\n", i, out_size);
+        fwrite(outbuf, 1, out_size, f);
     }
 
     /* add sequence end code to have a real mpeg file */
-    fwrite(endcode, 1, sizeof(endcode), f);
+    outbuf[0] = 0x00;
+    outbuf[1] = 0x00;
+    outbuf[2] = 0x01;
+    outbuf[3] = 0xb7;
+    fwrite(outbuf, 1, 4, f);
     fclose(f);
+    free(outbuf);
 
     avcodec_close(c);
     av_free(c);
-    av_freep(&frame->data[0]);
-    avcodec_free_frame(&frame);
+    av_free(picture->data[0]);
+    av_free(picture);
     printf("\n");
 }
 
@@ -488,42 +330,15 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
     fclose(f);
 }
 
-static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
-                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
-{
-    int len, got_frame;
-    char buf[1024];
-
-    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
-    if (len < 0) {
-        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
-        return len;
-    }
-    if (got_frame) {
-        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
-        fflush(stdout);
-
-        /* the picture is allocated by the decoder, no need to free it */
-        snprintf(buf, sizeof(buf), outfilename, *frame_count);
-        pgm_save(frame->data[0], frame->linesize[0],
-                 avctx->width, avctx->height, buf);
-        (*frame_count)++;
-    }
-    if (pkt->data) {
-        pkt->size -= len;
-        pkt->data += len;
-    }
-    return 0;
-}
-
 static void video_decode_example(const char *outfilename, const char *filename)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    int frame_count;
+    int frame, got_picture, len;
     FILE *f;
-    AVFrame *frame;
+    AVFrame *picture;
     uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    char buf[1024];
     AVPacket avpkt;
 
     av_init_packet(&avpkt);
@@ -531,20 +346,17 @@ static void video_decode_example(const char *outfilename, const char *filename)
     /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
     memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
 
-    printf("Decode video file %s to %s\n", filename, outfilename);
+    printf("Video decoding\n");
 
     /* find the mpeg1 video decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
+    codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
+    picture= avcodec_alloc_frame();
 
     if(codec->capabilities&CODEC_CAP_TRUNCATED)
         c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
@@ -554,24 +366,20 @@ static void video_decode_example(const char *outfilename, const char *filename)
        available in the bitstream. */
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
+    /* the codec gives us the frame size, in samples */
+
     f = fopen(filename, "rb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
 
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-
-    frame_count = 0;
+    frame = 0;
     for(;;) {
         avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
         if (avpkt.size == 0)
@@ -593,9 +401,26 @@ static void video_decode_example(const char *outfilename, const char *filename)
         /* here, we use a stream based decoder (mpeg1video), so we
            feed decoder and see if it could decode a frame */
         avpkt.data = inbuf;
-        while (avpkt.size > 0)
-            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
+        while (avpkt.size > 0) {
+            len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+            if (len < 0) {
+                fprintf(stderr, "Error while decoding frame %d\n", frame);
                 exit(1);
+            }
+            if (got_picture) {
+                printf("saving frame %3d\n", frame);
+                fflush(stdout);
+
+                /* the picture is allocated by the decoder. no need to
+                   free it */
+                snprintf(buf, sizeof(buf), outfilename, frame);
+                pgm_save(picture->data[0], picture->linesize[0],
+                         c->width, c->height, buf);
+                frame++;
+            }
+            avpkt.size -= len;
+            avpkt.data += len;
+        }
     }
 
     /* some codecs, such as MPEG, transmit the I and P frame with a
@@ -603,48 +428,50 @@ static void video_decode_example(const char *outfilename, const char *filename)
        chance to get the last frame of the video */
     avpkt.data = NULL;
     avpkt.size = 0;
-    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
+    len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+    if (got_picture) {
+        printf("saving last frame %3d\n", frame);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder. no need to
+           free it */
+        snprintf(buf, sizeof(buf), outfilename, frame);
+        pgm_save(picture->data[0], picture->linesize[0],
+                 c->width, c->height, buf);
+        frame++;
+    }
 
     fclose(f);
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&frame);
+    av_free(picture);
     printf("\n");
 }
 
 int main(int argc, char **argv)
 {
-    const char *output_type;
+    const char *filename;
+
+    /* must be called before using avcodec lib */
+    avcodec_init();
 
     /* register all the codecs */
     avcodec_register_all();
 
-    if (argc < 2) {
-        printf("usage: %s output_type\n"
-               "API example program to decode/encode a media stream with libavcodec.\n"
-               "This program generates a synthetic stream and encodes it to a file\n"
-               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
-               "The encoded stream is then decoded and written to a raw data output.\n"
-               "output_type must be choosen between 'h264', 'mp2', 'mpg'.\n",
-               argv[0]);
-        return 1;
-    }
-    output_type = argv[1];
+    if (argc <= 1) {
+        audio_encode_example("/tmp/test.mp2");
+        audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
 
-    if (!strcmp(output_type, "h264")) {
-        video_encode_example("test.h264", AV_CODEC_ID_H264);
-    } else if (!strcmp(output_type, "mp2")) {
-        audio_encode_example("test.mp2");
-        audio_decode_example("test.sw", "test.mp2");
-    } else if (!strcmp(output_type, "mpg")) {
-        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
-        video_decode_example("test%02d.pgm", "test.mpg");
+        video_encode_example("/tmp/test.h264", CODEC_ID_H264);
+        video_encode_example("/tmp/test.mpg", CODEC_ID_MPEG1VIDEO);
+        filename = "/tmp/test.mpg";
     } else {
-        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
-                output_type);
-        return 1;
+        filename = argv[1];
     }
 
+    //    audio_decode_example("/tmp/test.sw", filename);
+    video_decode_example("/tmp/test%d.pgm", filename);
+
     return 0;
 }

doc/examples/demuxing.c

@@ -1,340 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat demuxing API use example.
- *
- * Show how to use the libavformat and libavcodec API to demux and
- * decode audio and video data.
- * @example doc/examples/demuxing.c
- */
-
-#include <libavutil/imgutils.h>
-#include <libavutil/samplefmt.h>
-#include <libavutil/timestamp.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *fmt_ctx = NULL;
-static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
-static AVStream *video_stream = NULL, *audio_stream = NULL;
-static const char *src_filename = NULL;
-static const char *video_dst_filename = NULL;
-static const char *audio_dst_filename = NULL;
-static FILE *video_dst_file = NULL;
-static FILE *audio_dst_file = NULL;
-
-static uint8_t *video_dst_data[4] = {NULL};
-static int      video_dst_linesize[4];
-static int video_dst_bufsize;
-
-static uint8_t **audio_dst_data = NULL;
-static int       audio_dst_linesize;
-static int audio_dst_bufsize;
-
-static int video_stream_idx = -1, audio_stream_idx = -1;
-static AVFrame *frame = NULL;
-static AVPacket pkt;
-static int video_frame_count = 0;
-static int audio_frame_count = 0;
-
-static int decode_packet(int *got_frame, int cached)
-{
-    int ret = 0;
-
-    if (pkt.stream_index == video_stream_idx) {
-        /* decode video frame */
-        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame\n");
-            return ret;
-        }
-
-        if (*got_frame) {
-            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
-                   cached ? "(cached)" : "",
-                   video_frame_count++, frame->coded_picture_number,
-                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
-
-            /* copy decoded frame to destination buffer:
-             * this is required since rawvideo expects non aligned data */
-            av_image_copy(video_dst_data, video_dst_linesize,
-                          (const uint8_t **)(frame->data), frame->linesize,
-                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
-
-            /* write to rawvideo file */
-            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
-        }
-    } else if (pkt.stream_index == audio_stream_idx) {
-        /* decode audio frame */
-        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame\n");
-            return ret;
-        }
-
-        if (*got_frame) {
-            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
-                   cached ? "(cached)" : "",
-                   audio_frame_count++, frame->nb_samples,
-                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
-
-            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, frame->channels,
-                                   frame->nb_samples, frame->format, 1);
-            if (ret < 0) {
-                fprintf(stderr, "Could not allocate audio buffer\n");
-                return AVERROR(ENOMEM);
-            }
-
-            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
-            audio_dst_bufsize =
-                av_samples_get_buffer_size(NULL, frame->channels,
-                                           frame->nb_samples, frame->format, 1);
-
-            /* copy audio data to destination buffer:
-             * this is required since rawaudio expects non aligned data */
-            av_samples_copy(audio_dst_data, frame->data, 0, 0,
-                            frame->nb_samples, frame->channels, frame->format);
-
-            /* write to rawaudio file */
-            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
-            av_freep(&audio_dst_data[0]);
-        }
-    }
-
-    return ret;
-}
-
-static int open_codec_context(int *stream_idx,
-                              AVFormatContext *fmt_ctx, enum AVMediaType type)
-{
-    int ret;
-    AVStream *st;
-    AVCodecContext *dec_ctx = NULL;
-    AVCodec *dec = NULL;
-
-    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
-                av_get_media_type_string(type), src_filename);
-        return ret;
-    } else {
-        *stream_idx = ret;
-        st = fmt_ctx->streams[*stream_idx];
-
-        /* find decoder for the stream */
-        dec_ctx = st->codec;
-        dec = avcodec_find_decoder(dec_ctx->codec_id);
-        if (!dec) {
-            fprintf(stderr, "Failed to find %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-
-        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
-            fprintf(stderr, "Failed to open %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-static int get_format_from_sample_fmt(const char **fmt,
-                                      enum AVSampleFormat sample_fmt)
-{
-    int i;
-    struct sample_fmt_entry {
-        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
-    } sample_fmt_entries[] = {
-        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
-        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
-        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
-        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
-        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
-    };
-    *fmt = NULL;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
-        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
-        if (sample_fmt == entry->sample_fmt) {
-            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
-            return 0;
-        }
-    }
-
-    fprintf(stderr,
-            "sample format %s is not supported as output format\n",
-            av_get_sample_fmt_name(sample_fmt));
-    return -1;
-}
-
-int main (int argc, char **argv)
-{
-    int ret = 0, got_frame;
-
-    if (argc != 4) {
-        fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
-                "API example program to show how to read frames from an input file.\n"
-                "This program reads frames from a file, decodes them, and writes decoded\n"
-                "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n"
-                "\n", argv[0]);
-        exit(1);
-    }
-    src_filename = argv[1];
-    video_dst_filename = argv[2];
-    audio_dst_filename = argv[3];
-
-    /* register all formats and codecs */
-    av_register_all();
-
-    /* open input file, and allocate format context */
-    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
-        fprintf(stderr, "Could not open source file %s\n", src_filename);
-        exit(1);
-    }
-
-    /* retrieve stream information */
-    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        exit(1);
-    }
-
-    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
-        video_stream = fmt_ctx->streams[video_stream_idx];
-        video_dec_ctx = video_stream->codec;
-
-        video_dst_file = fopen(video_dst_filename, "wb");
-        if (!video_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
-            ret = 1;
-            goto end;
-        }
-
-        /* allocate image where the decoded image will be put */
-        ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             video_dec_ctx->width, video_dec_ctx->height,
-                             video_dec_ctx->pix_fmt, 1);
-        if (ret < 0) {
-            fprintf(stderr, "Could not allocate raw video buffer\n");
-            goto end;
-        }
-        video_dst_bufsize = ret;
-    }
-
-    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
-        int nb_planes;
-
-        audio_stream = fmt_ctx->streams[audio_stream_idx];
-        audio_dec_ctx = audio_stream->codec;
-        audio_dst_file = fopen(audio_dst_filename, "wb");
-        if (!audio_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
-            ret = 1;
-            goto end;
-        }
-
-        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
-            audio_dec_ctx->channels : 1;
-        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
-        if (!audio_dst_data) {
-            fprintf(stderr, "Could not allocate audio data buffers\n");
-            ret = AVERROR(ENOMEM);
-            goto end;
-        }
-    }
-
-    /* dump input information to stderr */
-    av_dump_format(fmt_ctx, 0, src_filename, 0);
-
-    if (!audio_stream && !video_stream) {
-        fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
-        ret = 1;
-        goto end;
-    }
-
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate frame\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    /* initialize packet, set data to NULL, let the demuxer fill it */
-    av_init_packet(&pkt);
-    pkt.data = NULL;
-    pkt.size = 0;
-
-    if (video_stream)
-        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
-    if (audio_stream)
-        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
-
-    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0)
-        decode_packet(&got_frame, 0);
-
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
-
-    printf("Demuxing succeeded.\n");
-
-    if (video_stream) {
-        printf("Play the output video file with the command:\n"
-               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-               av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
-               video_dst_filename);
-    }
-
-    if (audio_stream) {
-        const char *fmt;
-
-        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
-            goto end;
-        printf("Play the output audio file with the command:\n"
-               "ffplay -f %s -ac %d -ar %d %s\n",
-               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
-               audio_dst_filename);
-    }
-
-end:
-    if (video_dec_ctx)
-        avcodec_close(video_dec_ctx);
-    if (audio_dec_ctx)
-        avcodec_close(audio_dec_ctx);
-    avformat_close_input(&fmt_ctx);
-    if (video_dst_file)
-        fclose(video_dst_file);
-    if (audio_dst_file)
-        fclose(audio_dst_file);
-    av_free(frame);
-    av_free(video_dst_data[0]);
-    av_free(audio_dst_data);
-
-    return ret < 0;
-}

doc/examples/filtering.c

@@ -24,18 +24,14 @@
 /**
  * @file
  * API example for decoding and filtering
- * @example doc/examples/filtering_video.c
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
-#include <unistd.h>
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
+#include <libavfilter/vsrc_buffer.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -49,7 +45,7 @@ static int64_t last_pts = AV_NOPTS_VALUE;
 
 static int open_input_file(const char *filename)
 {
-    int ret;
+    int ret, i;
     AVCodec *dec;
 
     if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
@@ -57,7 +53,7 @@ static int open_input_file(const char *filename)
         return ret;
     }
 
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+    if ((ret = av_find_stream_info(fmt_ctx)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
         return ret;
     }
@@ -72,7 +68,7 @@ static int open_input_file(const char *filename)
     dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
 
     /* init the video decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+    if ((ret = avcodec_open(dec_ctx, dec)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
         return ret;
     }
@@ -85,21 +81,17 @@ static int init_filters(const char *filters_descr)
     char args[512];
     int ret;
     AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
-    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
-    AVBufferSinkParams *buffersink_params;
-
+    enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
     filter_graph = avfilter_graph_alloc();
 
     /* buffer video source: the decoded frames from the decoder will be inserted here. */
-    snprintf(args, sizeof(args),
-            "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
-            dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-            dec_ctx->time_base.num, dec_ctx->time_base.den,
-            dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
-
+    snprintf(args, sizeof(args), "%d:%d:%d:%d:%d:%d:%d",
+             dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
+             dec_ctx->time_base.num, dec_ctx->time_base.den,
+             dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
     ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                                        args, NULL, filter_graph);
     if (ret < 0) {
@@ -108,11 +100,8 @@ static int init_filters(const char *filters_descr)
     }
 
     /* buffer video sink: to terminate the filter chain. */
-    buffersink_params = av_buffersink_params_alloc();
-    buffersink_params->pixel_fmts = pix_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, buffersink_params, filter_graph);
-    av_free(buffersink_params);
+                                       NULL, pix_fmts, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
         return ret;
@@ -129,13 +118,12 @@ static int init_filters(const char *filters_descr)
     inputs->pad_idx    = 0;
     inputs->next       = NULL;
 
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
                                     &inputs, &outputs, NULL)) < 0)
         return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
         return ret;
-    return 0;
 }
 
 static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
@@ -200,42 +188,35 @@ int main(int argc, char **argv)
             avcodec_get_frame_defaults(&frame);
             got_frame = 0;
             ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
+            av_free_packet(&packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                 break;
             }
 
             if (got_frame) {
-                frame.pts = av_frame_get_best_effort_timestamp(&frame);
-
+                if (frame.pts == AV_NOPTS_VALUE)
+                    frame.pts = frame.pkt_dts == AV_NOPTS_VALUE ?
+                        frame.pkt_dts : frame.pkt_pts;
                 /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-                    break;
-                }
+                av_vsrc_buffer_add_frame(buffersrc_ctx, &frame);
 
                 /* pull filtered pictures from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if (ret < 0)
-                        goto end;
-
+                while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
+                    av_vsink_buffer_get_video_buffer_ref(buffersink_ctx, &picref, 0);
                     if (picref) {
                         display_picref(picref, buffersink_ctx->inputs[0]->time_base);
-                        avfilter_unref_bufferp(&picref);
+                        avfilter_unref_buffer(picref);
                     }
                 }
             }
         }
-        av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
     if (dec_ctx)
         avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
+    av_close_input_file(fmt_ctx);
 
     if (ret < 0 && ret != AVERROR_EOF) {
         char buf[1024];

doc/examples/filtering_audio.c

@@ -1,241 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2012 Clément Bœsch
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * API example for audio decoding and filtering
- * @example doc/examples/filtering_audio.c
- */
-
-#include <unistd.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-
-const char *filter_descr = "aresample=8000,aconvert=s16:mono";
-const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
-
-static AVFormatContext *fmt_ctx;
-static AVCodecContext *dec_ctx;
-AVFilterContext *buffersink_ctx;
-AVFilterContext *buffersrc_ctx;
-AVFilterGraph *filter_graph;
-static int audio_stream_index = -1;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *dec;
-
-    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    /* select the audio stream */
-    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
-        return ret;
-    }
-    audio_stream_index = ret;
-    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-
-    /* init the audio decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filters(const char *filters_descr)
-{
-    char args[512];
-    int ret;
-    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    AVFilter *abuffersink = avfilter_get_by_name("ffabuffersink");
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    AVABufferSinkParams *abuffersink_params;
-    const AVFilterLink *outlink;
-    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
-
-    filter_graph = avfilter_graph_alloc();
-
-    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (!dec_ctx->channel_layout)
-        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    snprintf(args, sizeof(args),
-            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
-             time_base.num, time_base.den, dec_ctx->sample_rate,
-             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
-    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
-                                       args, NULL, filter_graph);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
-    }
-
-    /* buffer audio sink: to terminate the filter chain. */
-    abuffersink_params = av_abuffersink_params_alloc();
-    abuffersink_params->sample_fmts     = sample_fmts;
-    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, abuffersink_params, filter_graph);
-    av_free(abuffersink_params);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        return ret;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
-                                    &inputs, &outputs, NULL)) < 0)
-        return ret;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-
-    /* Print summary of the sink buffer
-     * Note: args buffer is reused to store channel layout string */
-    outlink = buffersink_ctx->inputs[0];
-    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
-    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
-           (int)outlink->sample_rate,
-           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
-           args);
-
-    return 0;
-}
-
-static void print_samplesref(AVFilterBufferRef *samplesref)
-{
-    const AVFilterBufferRefAudioProps *props = samplesref->audio;
-    const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
-    const uint16_t *p     = (uint16_t*)samplesref->data[0];
-    const uint16_t *p_end = p + n;
-
-    while (p < p_end) {
-        fputc(*p    & 0xff, stdout);
-        fputc(*p>>8 & 0xff, stdout);
-        p++;
-    }
-    fflush(stdout);
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet;
-    AVFrame frame;
-    int got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
-        exit(1);
-    }
-
-    avcodec_register_all();
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = init_filters(filter_descr)) < 0)
-        goto end;
-
-    /* read all packets */
-    while (1) {
-        AVFilterBufferRef *samplesref;
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
-
-        if (packet.stream_index == audio_stream_index) {
-            avcodec_get_frame_defaults(&frame);
-            got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
-                continue;
-            }
-
-            if (got_frame) {
-                /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
-                    break;
-                }
-
-                /* pull filtered audio from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
-                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if(ret < 0)
-                        goto end;
-                    if (samplesref) {
-                        print_samplesref(samplesref);
-                        avfilter_unref_bufferp(&samplesref);
-                    }
-                }
-            }
-        }
-        av_free_packet(&packet);
-    }
-end:
-    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
-
-    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
-        exit(1);
-    }
-
-    exit(0);
-}

doc/examples/metadata.c

@@ -23,7 +23,6 @@
 /**
  * @file
  * Shows how the metadata API can be used in application programs.
- * @example doc/examples/metadata.c
  */
 
 #include <stdio.h>
@@ -51,6 +50,6 @@ int main (int argc, char **argv)
     while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
         printf("%s=%s\n", tag->key, tag->value);
 
-    avformat_close_input(&fmt_ctx);
+    avformat_free_context(fmt_ctx);
     return 0;
 }

doc/examples/muxing.c

@@ -26,7 +26,6 @@
  *
  * Output a media file in any supported libavformat format.
  * The default codecs are used.
- * @example doc/examples/muxing.c
  */
 
 #include <stdlib.h>
@@ -34,15 +33,17 @@
 #include <string.h>
 #include <math.h>
 
-#include <libavutil/mathematics.h>
-#include <libavformat/avformat.h>
-#include <libswscale/swscale.h>
+#include "libavutil/mathematics.h"
+#include "libavformat/avformat.h"
+#include "libswscale/swscale.h"
+
+#undef exit
 
 /* 5 seconds stream duration */
 #define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
 #define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
-#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
+#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */
 
 static int sws_flags = SWS_BICUBIC;
 
@@ -51,121 +52,93 @@ static int sws_flags = SWS_BICUBIC;
 
 static float t, tincr, tincr2;
 static int16_t *samples;
+static uint8_t *audio_outbuf;
+static int audio_outbuf_size;
 static int audio_input_frame_size;
 
-/* Add an output stream. */
-static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
-                            enum AVCodecID codec_id)
+/*
+ * add an audio output stream
+ */
+static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
     AVCodecContext *c;
     AVStream *st;
 
-    /* find the encoder */
-    *codec = avcodec_find_encoder(codec_id);
-    if (!(*codec)) {
-        fprintf(stderr, "Could not find encoder for '%s'\n",
-                avcodec_get_name(codec_id));
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, *codec);
+    st = avformat_new_stream(oc, NULL);
     if (!st) {
-        fprintf(stderr, "Could not allocate stream\n");
+        fprintf(stderr, "Could not alloc stream\n");
         exit(1);
     }
-    st->id = oc->nb_streams-1;
+    st->id = 1;
+
     c = st->codec;
+    c->codec_id = codec_id;
+    c->codec_type = AVMEDIA_TYPE_AUDIO;
 
-    switch ((*codec)->type) {
-    case AVMEDIA_TYPE_AUDIO:
-        st->id = 1;
-        c->sample_fmt  = AV_SAMPLE_FMT_S16;
-        c->bit_rate    = 64000;
-        c->sample_rate = 44100;
-        c->channels    = 2;
-        break;
+    /* put sample parameters */
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    c->bit_rate = 64000;
+    c->sample_rate = 44100;
+    c->channels = 2;
 
-    case AVMEDIA_TYPE_VIDEO:
-        avcodec_get_context_defaults3(c, *codec);
-        c->codec_id = codec_id;
-
-        c->bit_rate = 400000;
-        /* Resolution must be a multiple of two. */
-        c->width    = 352;
-        c->height   = 288;
-        /* timebase: This is the fundamental unit of time (in seconds) in terms
-         * of which frame timestamps are represented. For fixed-fps content,
-         * timebase should be 1/framerate and timestamp increments should be
-         * identical to 1. */
-        c->time_base.den = STREAM_FRAME_RATE;
-        c->time_base.num = 1;
-        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
-        c->pix_fmt       = STREAM_PIX_FMT;
-        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
-            /* just for testing, we also add B frames */
-            c->max_b_frames = 2;
-        }
-        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
-            /* Needed to avoid using macroblocks in which some coeffs overflow.
-             * This does not happen with normal video, it just happens here as
-             * the motion of the chroma plane does not match the luma plane. */
-            c->mb_decision = 2;
-        }
-    break;
-
-    default:
-        break;
-    }
-
-    /* Some formats want stream headers to be separate. */
+    // some formats want stream headers to be separate
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
         c->flags |= CODEC_FLAG_GLOBAL_HEADER;
 
     return st;
 }
 
-/**************************************************************/
-/* audio output */
-
-static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
-
-static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+static void open_audio(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
-    int ret;
+    AVCodec *codec;
 
     c = st->codec;
 
+    /* find the audio encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
     /* open it */
-    ret = avcodec_open2(c, codec, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     /* init signal generator */
-    t     = 0;
+    t = 0;
     tincr = 2 * M_PI * 110.0 / c->sample_rate;
     /* increment frequency by 110 Hz per second */
     tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        audio_input_frame_size = 10000;
-    else
+    audio_outbuf_size = 10000;
+    audio_outbuf = av_malloc(audio_outbuf_size);
+
+    /* ugly hack for PCM codecs (will be removed ASAP with new PCM
+       support to compute the input frame size in samples */
+    if (c->frame_size <= 1) {
+        audio_input_frame_size = audio_outbuf_size / c->channels;
+        switch(st->codec->codec_id) {
+        case CODEC_ID_PCM_S16LE:
+        case CODEC_ID_PCM_S16BE:
+        case CODEC_ID_PCM_U16LE:
+        case CODEC_ID_PCM_U16BE:
+            audio_input_frame_size >>= 1;
+            break;
+        default:
+            break;
+        }
+    } else {
         audio_input_frame_size = c->frame_size;
-    samples = av_malloc(audio_input_frame_size *
-                        av_get_bytes_per_sample(c->sample_fmt) *
-                        c->channels);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate audio samples buffer\n");
-        exit(1);
     }
+    samples = av_malloc(audio_input_frame_size * 2 * c->channels);
 }
 
-/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
- * 'nb_channels' channels. */
+/* prepare a 16 bit dummy audio frame of 'frame_size' samples and
+   'nb_channels' channels */
 static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
     int j, i, v;
@@ -174,9 +147,9 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
     q = samples;
     for (j = 0; j < frame_size; j++) {
         v = (int)(sin(t) * 10000);
-        for (i = 0; i < nb_channels; i++)
+        for(i = 0; i < nb_channels; i++)
             *q++ = v;
-        t     += tincr;
+        t += tincr;
         tincr += tincr2;
     }
 }
@@ -184,40 +157,26 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
-    AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet, ret;
-
+    AVPacket pkt;
     av_init_packet(&pkt);
+
     c = st->codec;
 
     get_audio_frame(samples, audio_input_frame_size, c->channels);
-    frame->nb_samples = audio_input_frame_size;
-    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             (uint8_t *)samples,
-                             audio_input_frame_size *
-                             av_get_bytes_per_sample(c->sample_fmt) *
-                             c->channels, 1);
 
-    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
-    if (ret < 0) {
-        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
-        exit(1);
-    }
-
-    if (!got_packet)
-        return;
+    pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);
 
+    if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
+        pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+    pkt.flags |= AV_PKT_FLAG_KEY;
     pkt.stream_index = st->index;
+    pkt.data = audio_outbuf;
 
-    /* Write the compressed frame to the media file. */
-    ret = av_interleaved_write_frame(oc, &pkt);
-    if (ret != 0) {
-        fprintf(stderr, "Error while writing audio frame: %s\n",
-                av_err2str(ret));
+    /* write the compressed frame in the media file */
+    if (av_interleaved_write_frame(oc, &pkt) != 0) {
+        fprintf(stderr, "Error while writing audio frame\n");
         exit(1);
     }
-    avcodec_free_frame(&frame);
 }
 
 static void close_audio(AVFormatContext *oc, AVStream *st)
@@ -225,73 +184,160 @@ static void close_audio(AVFormatContext *oc, AVStream *st)
     avcodec_close(st->codec);
 
     av_free(samples);
+    av_free(audio_outbuf);
 }
 
 /**************************************************************/
 /* video output */
 
-static AVFrame *frame;
-static AVPicture src_picture, dst_picture;
-static int frame_count;
+static AVFrame *picture, *tmp_picture;
+static uint8_t *video_outbuf;
+static int frame_count, video_outbuf_size;
 
-static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+/* add a video output stream */
+static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
-    int ret;
-    AVCodecContext *c = st->codec;
+    AVCodecContext *c;
+    AVStream *st;
+    AVCodec *codec;
+
+    st = avformat_new_stream(oc, NULL);
+    if (!st) {
+        fprintf(stderr, "Could not alloc stream\n");
+        exit(1);
+    }
+
+    c = st->codec;
+
+    /* find the video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+    avcodec_get_context_defaults3(c, codec);
+
+    c->codec_id = codec_id;
+
+    /* put sample parameters */
+    c->bit_rate = 400000;
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* time base: this is the fundamental unit of time (in seconds) in terms
+       of which frame timestamps are represented. for fixed-fps content,
+       timebase should be 1/framerate and timestamp increments should be
+       identically 1. */
+    c->time_base.den = STREAM_FRAME_RATE;
+    c->time_base.num = 1;
+    c->gop_size = 12; /* emit one intra frame every twelve frames at most */
+    c->pix_fmt = STREAM_PIX_FMT;
+    if (c->codec_id == CODEC_ID_MPEG2VIDEO) {
+        /* just for testing, we also add B frames */
+        c->max_b_frames = 2;
+    }
+    if (c->codec_id == CODEC_ID_MPEG1VIDEO){
+        /* Needed to avoid using macroblocks in which some coeffs overflow.
+           This does not happen with normal video, it just happens here as
+           the motion of the chroma plane does not match the luma plane. */
+        c->mb_decision=2;
+    }
+    // some formats want stream headers to be separate
+    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    return st;
+}
+
+static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
+{
+    AVFrame *picture;
+    uint8_t *picture_buf;
+    int size;
+
+    picture = avcodec_alloc_frame();
+    if (!picture)
+        return NULL;
+    size = avpicture_get_size(pix_fmt, width, height);
+    picture_buf = av_malloc(size);
+    if (!picture_buf) {
+        av_free(picture);
+        return NULL;
+    }
+    avpicture_fill((AVPicture *)picture, picture_buf,
+                   pix_fmt, width, height);
+    return picture;
+}
+
+static void open_video(AVFormatContext *oc, AVStream *st)
+{
+    AVCodec *codec;
+    AVCodecContext *c;
+
+    c = st->codec;
+
+    /* find the video encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
 
     /* open the codec */
-    ret = avcodec_open2(c, codec, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
-    /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
+    video_outbuf = NULL;
+    if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
+        /* allocate output buffer */
+        /* XXX: API change will be done */
+        /* buffers passed into lav* can be allocated any way you prefer,
+           as long as they're aligned enough for the architecture, and
+           they're freed appropriately (such as using av_free for buffers
+           allocated with av_malloc) */
+        video_outbuf_size = 200000;
+        video_outbuf = av_malloc(video_outbuf_size);
+    }
+
+    /* allocate the encoded raw picture */
+    picture = alloc_picture(c->pix_fmt, c->width, c->height);
+    if (!picture) {
+        fprintf(stderr, "Could not allocate picture\n");
         exit(1);
     }
 
-    /* Allocate the encoded raw picture. */
-    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
-        exit(1);
-    }
-
-    /* If the output format is not YUV420P, then a temporary YUV420P
-     * picture is needed too. It is then converted to the required
-     * output format. */
-    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
-        if (ret < 0) {
-            fprintf(stderr, "Could not allocate temporary picture: %s\n",
-                    av_err2str(ret));
+    /* if the output format is not YUV420P, then a temporary YUV420P
+       picture is needed too. It is then converted to the required
+       output format */
+    tmp_picture = NULL;
+    if (c->pix_fmt != PIX_FMT_YUV420P) {
+        tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
+        if (!tmp_picture) {
+            fprintf(stderr, "Could not allocate temporary picture\n");
             exit(1);
         }
     }
-
-    /* copy data and linesize picture pointers to frame */
-    *((AVPicture *)frame) = dst_picture;
 }
 
-/* Prepare a dummy image. */
-static void fill_yuv_image(AVPicture *pict, int frame_index,
-                           int width, int height)
+/* prepare a dummy image */
+static void fill_yuv_image(AVFrame *pict, int frame_index, int width, int height)
 {
     int x, y, i;
 
     i = frame_index;
 
     /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
+    for (y = 0; y < height; y++) {
+        for (x = 0; x < width; x++) {
             pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+        }
+    }
 
     /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
+    for (y = 0; y < height/2; y++) {
+        for (x = 0; x < width/2; x++) {
             pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
             pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
         }
@@ -300,78 +346,76 @@ static void fill_yuv_image(AVPicture *pict, int frame_index,
 
 static void write_video_frame(AVFormatContext *oc, AVStream *st)
 {
-    int ret;
-    static struct SwsContext *sws_ctx;
-    AVCodecContext *c = st->codec;
+    int out_size, ret;
+    AVCodecContext *c;
+    static struct SwsContext *img_convert_ctx;
+
+    c = st->codec;
 
     if (frame_count >= STREAM_NB_FRAMES) {
-        /* No more frames to compress. The codec has a latency of a few
-         * frames if using B-frames, so we get the last frames by
-         * passing the same picture again. */
+        /* no more frame to compress. The codec has a latency of a few
+           frames if using B frames, so we get the last frames by
+           passing the same picture again */
     } else {
-        if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+        if (c->pix_fmt != PIX_FMT_YUV420P) {
             /* as we only generate a YUV420P picture, we must convert it
-             * to the codec pixel format if needed */
-            if (!sws_ctx) {
-                sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
-                                         c->width, c->height, c->pix_fmt,
-                                         sws_flags, NULL, NULL, NULL);
-                if (!sws_ctx) {
-                    fprintf(stderr,
-                            "Could not initialize the conversion context\n");
+               to the codec pixel format if needed */
+            if (img_convert_ctx == NULL) {
+                img_convert_ctx = sws_getContext(c->width, c->height,
+                                                 PIX_FMT_YUV420P,
+                                                 c->width, c->height,
+                                                 c->pix_fmt,
+                                                 sws_flags, NULL, NULL, NULL);
+                if (img_convert_ctx == NULL) {
+                    fprintf(stderr, "Cannot initialize the conversion context\n");
                     exit(1);
                 }
             }
-            fill_yuv_image(&src_picture, frame_count, c->width, c->height);
-            sws_scale(sws_ctx,
-                      (const uint8_t * const *)src_picture.data, src_picture.linesize,
-                      0, c->height, dst_picture.data, dst_picture.linesize);
+            fill_yuv_image(tmp_picture, frame_count, c->width, c->height);
+            sws_scale(img_convert_ctx, tmp_picture->data, tmp_picture->linesize,
+                      0, c->height, picture->data, picture->linesize);
         } else {
-            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
+            fill_yuv_image(picture, frame_count, c->width, c->height);
         }
     }
 
+
     if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* Raw video case - directly store the picture in the packet */
+        /* raw video case. The API will change slightly in the near
+           future for that. */
         AVPacket pkt;
         av_init_packet(&pkt);
 
-        pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = st->index;
-        pkt.data          = dst_picture.data[0];
-        pkt.size          = sizeof(AVPicture);
+        pkt.flags |= AV_PKT_FLAG_KEY;
+        pkt.stream_index = st->index;
+        pkt.data = (uint8_t *)picture;
+        pkt.size = sizeof(AVPicture);
 
         ret = av_interleaved_write_frame(oc, &pkt);
     } else {
         /* encode the image */
-        AVPacket pkt;
-        int got_output;
+        out_size = avcodec_encode_video(c, video_outbuf, video_outbuf_size, picture);
+        /* if zero size, it means the image was buffered */
+        if (out_size > 0) {
+            AVPacket pkt;
+            av_init_packet(&pkt);
 
-        av_init_packet(&pkt);
-        pkt.data = NULL;    // packet data will be allocated by the encoder
-        pkt.size = 0;
-
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
-            exit(1);
-        }
-
-        /* If size is zero, it means the image was buffered. */
-        if (got_output) {
-            if (c->coded_frame->key_frame)
+            if (c->coded_frame->pts != AV_NOPTS_VALUE)
+                pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+            if(c->coded_frame->key_frame)
                 pkt.flags |= AV_PKT_FLAG_KEY;
-
             pkt.stream_index = st->index;
+            pkt.data = video_outbuf;
+            pkt.size = out_size;
 
-            /* Write the compressed frame to the media file. */
+            /* write the compressed frame in the media file */
             ret = av_interleaved_write_frame(oc, &pkt);
         } else {
             ret = 0;
         }
     }
     if (ret != 0) {
-        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+        fprintf(stderr, "Error while writing video frame\n");
         exit(1);
     }
     frame_count++;
@@ -380,9 +424,13 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
 static void close_video(AVFormatContext *oc, AVStream *st)
 {
     avcodec_close(st->codec);
-    av_free(src_picture.data[0]);
-    av_free(dst_picture.data[0]);
-    av_free(frame);
+    av_free(picture->data[0]);
+    av_free(picture);
+    if (tmp_picture) {
+        av_free(tmp_picture->data[0]);
+        av_free(tmp_picture);
+    }
+    av_free(video_outbuf);
 }
 
 /**************************************************************/
@@ -394,20 +442,17 @@ int main(int argc, char **argv)
     AVOutputFormat *fmt;
     AVFormatContext *oc;
     AVStream *audio_st, *video_st;
-    AVCodec *audio_codec, *video_codec;
     double audio_pts, video_pts;
-    int ret, i;
+    int i;
 
-    /* Initialize libavcodec, and register all codecs and formats. */
+    /* initialize libavcodec, and register all codecs and formats */
     av_register_all();
 
     if (argc != 2) {
         printf("usage: %s output_file\n"
                "API example program to output a media file with libavformat.\n"
-               "This program generates a synthetic audio and video stream, encodes and\n"
-               "muxes them into a file named output_file.\n"
                "The output format is automatically guessed according to the file extension.\n"
-               "Raw images can also be output by using '%%d' in the filename.\n"
+               "Raw images can also be output by using '%%d' in the filename\n"
                "\n", argv[0]);
         return 1;
     }
@@ -425,57 +470,46 @@ int main(int argc, char **argv)
     }
     fmt = oc->oformat;
 
-    /* Add the audio and video streams using the default format codecs
-     * and initialize the codecs. */
+    /* add the audio and video streams using the default format codecs
+       and initialize the codecs */
     video_st = NULL;
     audio_st = NULL;
-
-    if (fmt->video_codec != AV_CODEC_ID_NONE) {
-        video_st = add_stream(oc, &video_codec, fmt->video_codec);
+    if (fmt->video_codec != CODEC_ID_NONE) {
+        video_st = add_video_stream(oc, fmt->video_codec);
     }
-    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
-        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
+    if (fmt->audio_codec != CODEC_ID_NONE) {
+        audio_st = add_audio_stream(oc, fmt->audio_codec);
     }
 
-    /* Now that all the parameters are set, we can open the audio and
-     * video codecs and allocate the necessary encode buffers. */
-    if (video_st)
-        open_video(oc, video_codec, video_st);
-    if (audio_st)
-        open_audio(oc, audio_codec, audio_st);
-
     av_dump_format(oc, 0, filename, 1);
 
+    /* now that all the parameters are set, we can open the audio and
+       video codecs and allocate the necessary encode buffers */
+    if (video_st)
+        open_video(oc, video_st);
+    if (audio_st)
+        open_audio(oc, audio_st);
+
     /* open the output file, if needed */
     if (!(fmt->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            fprintf(stderr, "Could not open '%s': %s\n", filename,
-                    av_err2str(ret));
+        if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
+            fprintf(stderr, "Could not open '%s'\n", filename);
             return 1;
         }
     }
 
-    /* Write the stream header, if any. */
-    ret = avformat_write_header(oc, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred when opening output file: %s\n",
-                av_err2str(ret));
-        return 1;
-    }
-
-    if (frame)
-        frame->pts = 0;
-    for (;;) {
-        /* Compute current audio and video time. */
+    /* write the stream header, if any */
+    av_write_header(oc);
+    picture->pts = 0;
+    for(;;) {
+        /* compute current audio and video time */
         if (audio_st)
             audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
         else
             audio_pts = 0.0;
 
         if (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
+            video_pts = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
         else
             video_pts = 0.0;
 
@@ -488,31 +522,32 @@ int main(int argc, char **argv)
             write_audio_frame(oc, audio_st);
         } else {
             write_video_frame(oc, video_st);
-            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
+            picture->pts++;
         }
     }
 
-    /* Write the trailer, if any. The trailer must be written before you
-     * close the CodecContexts open when you wrote the header; otherwise
-     * av_write_trailer() may try to use memory that was freed on
-     * av_codec_close(). */
+    /* write the trailer, if any.  the trailer must be written
+     * before you close the CodecContexts open when you wrote the
+     * header; otherwise write_trailer may try to use memory that
+     * was freed on av_codec_close() */
     av_write_trailer(oc);
 
-    /* Close each codec. */
+    /* close each codec */
     if (video_st)
         close_video(oc, video_st);
     if (audio_st)
         close_audio(oc, audio_st);
 
-    /* Free the streams. */
-    for (i = 0; i < oc->nb_streams; i++) {
+    /* free the streams */
+    for(i = 0; i < oc->nb_streams; i++) {
         av_freep(&oc->streams[i]->codec);
         av_freep(&oc->streams[i]);
     }
 
-    if (!(fmt->flags & AVFMT_NOFILE))
-        /* Close the output file. */
+    if (!(fmt->flags & AVFMT_NOFILE)) {
+        /* close the output file */
         avio_close(oc->pb);
+    }
 
     /* free the stream */
     av_free(oc);

doc/examples/resampling_audio.c

@@ -1,223 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @example doc/examples/resampling_audio.c
- * libswresample API use example.
- */
-
-#include <libavutil/opt.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/samplefmt.h>
-#include <libswresample/swresample.h>
-
-static int get_format_from_sample_fmt(const char **fmt,
-                                      enum AVSampleFormat sample_fmt)
-{
-    int i;
-    struct sample_fmt_entry {
-        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
-    } sample_fmt_entries[] = {
-        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
-        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
-        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
-        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
-        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
-    };
-    *fmt = NULL;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
-        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
-        if (sample_fmt == entry->sample_fmt) {
-            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
-            return 0;
-        }
-    }
-
-    fprintf(stderr,
-            "Sample format %s not supported as output format\n",
-            av_get_sample_fmt_name(sample_fmt));
-    return AVERROR(EINVAL);
-}
-
-/**
- * Fill dst buffer with nb_samples, generated starting from t.
- */
-void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
-{
-    int i, j;
-    double tincr = 1.0 / sample_rate, *dstp = dst;
-    const double c = 2 * M_PI * 440.0;
-
-    /* generate sin tone with 440Hz frequency and duplicated channels */
-    for (i = 0; i < nb_samples; i++) {
-        *dstp = sin(c * *t);
-        for (j = 1; j < nb_channels; j++)
-            dstp[j] = dstp[0];
-        dstp += nb_channels;
-        *t += tincr;
-    }
-}
-
-int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
-                                    int nb_samples, enum AVSampleFormat sample_fmt, int align)
-{
-    int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
-
-    *data = av_malloc(sizeof(*data) * nb_planes);
-    if (!*data)
-        return AVERROR(ENOMEM);
-    return av_samples_alloc(*data, linesize, nb_channels,
-                            nb_samples, sample_fmt, align);
-}
-
-int main(int argc, char **argv)
-{
-    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
-    int src_rate = 48000, dst_rate = 44100;
-    uint8_t **src_data = NULL, **dst_data = NULL;
-    int src_nb_channels = 0, dst_nb_channels = 0;
-    int src_linesize, dst_linesize;
-    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
-    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
-    const char *dst_filename = NULL;
-    FILE *dst_file;
-    int dst_bufsize;
-    const char *fmt;
-    struct SwrContext *swr_ctx;
-    double t;
-    int ret;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s output_file\n"
-                "API example program to show how to resample an audio stream with libswresample.\n"
-                "This program generates a series of audio frames, resamples them to a specified "
-                "output format and rate and saves them to an output file named output_file.\n",
-            argv[0]);
-        exit(1);
-    }
-    dst_filename = argv[1];
-
-    dst_file = fopen(dst_filename, "wb");
-    if (!dst_file) {
-        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
-        exit(1);
-    }
-
-    /* create resampler context */
-    swr_ctx = swr_alloc();
-    if (!swr_ctx) {
-        fprintf(stderr, "Could not allocate resampler context\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    /* set options */
-    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
-    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
-    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
-
-    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
-    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
-    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
-
-    /* initialize the resampling context */
-    if ((ret = swr_init(swr_ctx)) < 0) {
-        fprintf(stderr, "Failed to initialize the resampling context\n");
-        goto end;
-    }
-
-    /* allocate source and destination samples buffers */
-
-    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
-    ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
-                                       src_nb_samples, src_sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate source samples\n");
-        goto end;
-    }
-
-    /* compute the number of converted samples: buffering is avoided
-     * ensuring that the output buffer will contain at least all the
-     * converted input samples */
-    max_dst_nb_samples = dst_nb_samples =
-        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
-
-    /* buffer is going to be directly written to a rawaudio file, no alignment */
-    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
-    ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
-                                       dst_nb_samples, dst_sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate destination samples\n");
-        goto end;
-    }
-
-    t = 0;
-    do {
-        /* generate synthetic audio */
-        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
-
-        /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
-                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
-        if (dst_nb_samples > max_dst_nb_samples) {
-            av_free(dst_data[0]);
-            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
-                                   dst_nb_samples, dst_sample_fmt, 1);
-            if (ret < 0)
-                break;
-            max_dst_nb_samples = dst_nb_samples;
-        }
-
-        /* convert to destination format */
-        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            goto end;
-        }
-        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
-                                                 ret, dst_sample_fmt, 1);
-        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
-        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
-    } while (t < 10);
-
-    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
-        goto end;
-    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
-            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
-            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
-
-end:
-    if (dst_file)
-        fclose(dst_file);
-
-    if (src_data)
-        av_freep(&src_data[0]);
-    av_freep(&src_data);
-
-    if (dst_data)
-        av_freep(&dst_data[0]);
-    av_freep(&dst_data);
-
-    swr_free(&swr_ctx);
-    return ret < 0;
-}

doc/examples/scaling_video.c

@@ -1,141 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libswscale API use example.
- * @example doc/examples/scaling_video.c
- */
-
-#include <libavutil/imgutils.h>
-#include <libavutil/parseutils.h>
-#include <libswscale/swscale.h>
-
-static void fill_yuv_image(uint8_t *data[4], int linesize[4],
-                           int width, int height, int frame_index)
-{
-    int x, y;
-
-    /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
-            data[0][y * linesize[0] + x] = x + y + frame_index * 3;
-
-    /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
-            data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
-            data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
-        }
-    }
-}
-
-int main(int argc, char **argv)
-{
-    uint8_t *src_data[4], *dst_data[4];
-    int src_linesize[4], dst_linesize[4];
-    int src_w = 320, src_h = 240, dst_w, dst_h;
-    enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
-    const char *dst_size = NULL;
-    const char *dst_filename = NULL;
-    FILE *dst_file;
-    int dst_bufsize;
-    struct SwsContext *sws_ctx;
-    int i, ret;
-
-    if (argc != 3) {
-        fprintf(stderr, "Usage: %s output_file output_size\n"
-                "API example program to show how to scale an image with libswscale.\n"
-                "This program generates a series of pictures, rescales them to the given "
-                "output_size and saves them to an output file named output_file\n."
-                "\n", argv[0]);
-        exit(1);
-    }
-    dst_filename = argv[1];
-    dst_size     = argv[2];
-
-    if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
-        fprintf(stderr,
-                "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
-                dst_size);
-        exit(1);
-    }
-
-    dst_file = fopen(dst_filename, "wb");
-    if (!dst_file) {
-        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
-        exit(1);
-    }
-
-    /* create scaling context */
-    sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
-                             dst_w, dst_h, dst_pix_fmt,
-                             SWS_BILINEAR, NULL, NULL, NULL);
-    if (!sws_ctx) {
-        fprintf(stderr,
-                "Impossible to create scale context for the conversion "
-                "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
-                av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
-                av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
-        ret = AVERROR(EINVAL);
-        goto end;
-    }
-
-    /* allocate source and destination image buffers */
-    if ((ret = av_image_alloc(src_data, src_linesize,
-                              src_w, src_h, src_pix_fmt, 16)) < 0) {
-        fprintf(stderr, "Could not allocate source image\n");
-        goto end;
-    }
-
-    /* buffer is going to be written to rawvideo file, no alignmnet */
-    if ((ret = av_image_alloc(dst_data, dst_linesize,
-                              dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
-        fprintf(stderr, "Could not allocate destination image\n");
-        goto end;
-    }
-    dst_bufsize = ret;
-
-    for (i = 0; i < 100; i++) {
-        /* generate synthetic video */
-        fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
-
-        /* convert to destination format */
-        sws_scale(sws_ctx, (const uint8_t * const*)src_data,
-                  src_linesize, 0, src_h, dst_data, dst_linesize);
-
-        /* write scaled image to file */
-        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
-    }
-
-    fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
-           "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-           av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
-
-end:
-    if (dst_file)
-        fclose(dst_file);
-    av_freep(&src_data[0]);
-    av_freep(&dst_data[0]);
-    sws_freeContext(sws_ctx);
-    return ret < 0;
-}

doc/faq.texi

@@ -79,17 +79,6 @@ not a bug they should fix:
 Then again, some of them do not know the difference between an undecidable
 problem and an NP-hard problem...
 
-@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
-
-Distributions usually split libraries in several packages. The main package
-contains the files necessary to run programs using the library. The
-development package contains the files necessary to build programs using the
-library. Sometimes, docs and/or data are in a separate package too.
-
-To build FFmpeg, you need to install the development package. It is usually
-called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
-build is finished, but be sure to keep the main package.
-
 @chapter Usage
 
 @section ffmpeg does not work; what is wrong?
@@ -110,16 +99,7 @@ Then you may run:
 
 Notice that @samp{%d} is replaced by the image number.
 
-@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
-
-Use the @option{-start_number} option to declare a starting number for
-the sequence. This is useful if your sequence does not start with
-@file{img001.jpg} but is still in a numerical order. The following
-example will start with @file{img100.jpg}:
-
-@example
-  ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
-@end example
+@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
@@ -142,12 +122,6 @@ Then run:
 
 The same logic is used for any image format that ffmpeg reads.
 
-You can also use @command{cat} to pipe images to ffmpeg:
-
-@example
-  cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
-@end example
-
 @section How do I encode movie to single pictures?
 
 Use:
@@ -239,63 +213,8 @@ For ANY other help on Avisynth, please visit the
 
 @section How can I join video files?
 
-To "join" video files is quite ambiguous. The following list explains the
-different kinds of "joining" and points out how those are addressed in
-FFmpeg. To join video files may mean:
-
-@itemize
-
-@item
-To put them one after the other: this is called to @emph{concatenate} them
-(in short: concat) and is addressed
-@ref{How can I concatenate video files, in this very faq}.
-
-@item
-To put them together in the same file, to let the user choose between the
-different versions (example: different audio languages): this is called to
-@emph{multiplex} them together (in short: mux), and is done by simply
-invoking ffmpeg with several @option{-i} options.
-
-@item
-For audio, to put all channels together in a single stream (example: two
-mono streams into one stereo stream): this is sometimes called to
-@emph{merge} them, and can be done using the
-@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
-
-@item
-For audio, to play one on top of the other: this is called to @emph{mix}
-them, and can be done by first merging them into a single stream and then
-using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
-the channels at will.
-
-@item
-For video, to display both together, side by side or one on top of a part of
-the other; it can be done using the
-@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
-
-@end itemize
-
-@anchor{How can I concatenate video files}
-@section How can I concatenate video files?
-
-There are several solutions, depending on the exact circumstances.
-
-@subsection Concatenating using the concat @emph{filter}
-
-FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
-@code{concat}} filter designed specifically for that, with examples in the
-documentation. This operation is recommended if you need to re-encode.
-
-@subsection Concatenating using the concat @emph{demuxer}
-
-FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
-@code{concat}} demuxer which you can use when you want to avoid a re-encode and
-your format doesn't support file level concatenation.
-
-@subsection Concatenating using the concat @emph{protocol} (file level)
-
-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
-video by merely concatenating the files them.
+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
@@ -303,38 +222,28 @@ equally humble @code{copy} under Windows), and finally transcoding back to your
 format of choice.
 
 @example
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
-ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+ffmpeg -i input1.avi -same_quant intermediate1.mpg
+ffmpeg -i input2.avi -same_quant intermediate2.mpg
 cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+ffmpeg -i intermediate_all.mpg -same_quant output.avi
 @end example
 
-Additionally, you can use the @code{concat} protocol instead of @code{cat} or
-@code{copy} which will avoid creation of a potentially huge intermediate file.
+Notice that you should either use @code{-same_quant} or set a reasonably high
+bitrate for your intermediate and output files, if you want to preserve
+video quality.
 
-@example
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
-ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
-ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
-@end example
-
-Note that you may need to escape the character "|" which is special for many
-shells.
-
-Another option is usage of named pipes, should your platform support it:
+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 -qscale:v 1 -y intermediate1.mpg < /dev/null &
-ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
+ffmpeg -i input1.avi -same_quant -y intermediate1.mpg < /dev/null &
+ffmpeg -i input2.avi -same_quant -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
+ffmpeg -f mpeg -i - -same_quant -c:v mpeg4 -acodec libmp3lame output.avi
 @end example
 
-@subsection Concatenating using raw audio and video
-
 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
@@ -342,8 +251,7 @@ 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 concatenate two FLV files into an
-output.flv file:
+For example, let's say we want to join two FLV files into an output.flv file:
 
 @example
 mkfifo temp1.a
@@ -360,7 +268,7 @@ 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 \
-       -y output.flv
+       -same_quant -y output.flv
 rm temp[12].[av] all.[av]
 @end example
 
@@ -402,48 +310,11 @@ specifying the exact format.
 aconvert=s16:stereo:packed
 @end example
 
-@section Why does FFmpeg not see the subtitles in my VOB file?
-
-VOB and a few other formats do not have a global header that describes
-everything present in the file. Instead, applications are supposed to scan
-the file to see what it contains. Since VOB files are frequently large, only
-the beginning is scanned. If the subtitles happen only later in the file,
-they will not be initally detected.
-
-Some applications, including the @code{ffmpeg} command-line tool, can only
-work with streams that were detected during the initial scan; streams that
-are detected later are ignored.
-
-The size of the initial scan is controlled by two options: @code{probesize}
-(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
-the subtitle stream to be detected, both values must be large enough.
-
-@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
-
-The @option{-sameq} option meant "same quantizer", and made sense only in a
-very limited set of cases. Unfortunately, a lot of people mistook it for
-"same quality" and used it in places where it did not make sense: it had
-roughly the expected visible effect, but achieved it in a very inefficient
-way.
-
-Each encoder has its own set of options to set the quality-vs-size balance,
-use the options for the encoder you are using to set the quality level to a
-point acceptable for your tastes. The most common options to do that are
-@option{-qscale} and @option{-qmax}, but you should peruse the documentation
-of the encoder you chose.
-
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
 
-Yes. Check the @file{doc/examples} directory in the source
-repository, also available online at:
-@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
-
-Examples are also installed by default, usually in
-@code{$PREFIX/share/ffmpeg/examples}.
-
-Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
+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}).
 
@@ -455,8 +326,31 @@ with @code{#ifdef}s related to the compiler.
 
 @section Is Microsoft Visual C++ supported?
 
-Yes. Please see the @uref{platform.html, Microsoft Visual C++}
-section in the FFmpeg documentation.
+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?
 
@@ -481,24 +375,6 @@ 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'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.
-
-FFmpeg builds static libraries by default. In static libraries, dependencies
-are not handled. That has two consequences. First, you must specify the
-libraries in dependency order: @code{-lavdevice} must come before
-@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
-Second, external libraries that are used in FFmpeg have to be specified too.
-
-An easy way to get the full list of required libraries in dependency order
-is to use @code{pkg-config}.
-
-@example
-  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
-@end example
-
-See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
-more details.
-
 @section I'm using FFmpeg 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
@@ -514,8 +390,8 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
 
 @section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
 
-You have to create a custom AVIOContext using @code{avio_alloc_context},
-see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
+You have to implement a URLProtocol, see @file{libavformat/file.c} in
+FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
 
 @section Where can I find libav* headers for Pascal/Delphi?
 

doc/fate.texi

@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle FFmpeg Automated Testing Environment
+@settitle FATE Automated Testing Environment
 @titlepage
-@center @titlefont{FFmpeg Automated Testing Environment}
+@center @titlefont{FATE Automated Testing Environment}
 @end titlepage
 
 @node Top
@@ -78,14 +78,11 @@ Do not put a '~' character in the samples path to indicate a home
 directory. Because of shell nuances, this will cause FATE to fail.
 @end float
 
-To use a custom wrapper to run the test, pass @option{--target-exec} to
-@command{configure} or set the @var{TARGET_EXEC} Make variable.
-
 
 @chapter Submitting the results to the FFmpeg result aggregation server
 
   To submit your results to the server you should run fate through the
-shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
+shell script tests/fate.sh from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.
 
 @example
@@ -93,11 +90,11 @@ tests/fate.sh /path/to/fate_config
 @end example
 
   A configuration file template with comments describing the individual
-configuration variables can be found at @file{doc/fate_config.sh.template}.
+configuration variables can be found at @file{tests/fate_config.sh.template}.
 
 @ifhtml
   The mentioned configuration template is also available here:
-@verbatiminclude fate_config.sh.template
+@verbatiminclude ../tests/fate_config.sh.template
 @end ifhtml
 
   Create a configuration that suits your needs, based on the configuration
@@ -121,9 +118,8 @@ present in $workdir as specified in the configuration file:
     @item version
 @end itemize
 
-  When you have everything working properly you can create an SSH key pair
-and send the public key to the FATE server administrator who can be contacted
-at the email address @email{fate-admin@@ffmpeg.org}.
+  When you have everything working properly you can create an SSH key and
+send its public part to the FATE server administrator.
 
   Configure your SSH client to use public key authentication with that key
 when connecting to the FATE server. Also do not forget to check the identity
@@ -133,11 +129,6 @@ The FATE server's fingerprint is:
 
   b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
 
-  If you have problems connecting to the FATE server, it may help to try out
-the @command{ssh} command with one or more @option{-v} options. You should
-get detailed output concerning your SSH configuration and the authentication
-process.
-
   The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.
 
@@ -175,20 +166,9 @@ the synchronisation of the samples directory.
 @item THREADS
     Specify how many threads to use while running regression tests, it is
     quite useful to detect thread-related regressions.
-@item THREAD_TYPE
-    Specify which threading strategy test, either @var{slice} or @var{frame},
-    by default @var{slice+frame}
-@item CPUFLAGS
-    Specify CPU flags.
-@item TARGET_EXEC
-    Specify or override the wrapper used to run the tests.
-    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
-    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-    through @command{ssh}.
 @end table
 
-@section Examples
-
+Example:
 @example
-make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
+make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
 @end example

doc/ffmpeg-bitstream-filters.texi

@@ -1,45 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Bitstream Filters Documentation
-@titlepage
-@center @titlefont{FFmpeg Bitstream Filters Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the bitstream filters provided by the
-libavcodec library.
-
-A bitstream filter operates on the encoded stream data, and performs
-bitstream level modifications without performing decoding.
-
-@c man end DESCRIPTION
-
-@include bitstream_filters.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavcodec.html,libavcodec}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-bitstream-filters
-@settitle FFmpeg bitstream filters
-
-@end ignore
-
-@bye

doc/ffmpeg-devices.texi

@@ -1,62 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Devices Documentation
-@titlepage
-@center @titlefont{FFmpeg Devices Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the input and output devices provided by the
-libavdevice library.
-
-@c man end DESCRIPTION
-
-@chapter Device Options
-@c man begin DEVICE OPTIONS
-
-The libavdevice library provides the same interface as
-libavformat. Namely, an input device is considered like a demuxer, and
-an output device like a muxer, and the interface and generic device
-options are the same provided by libavformat (see the ffmpeg-formats
-manual).
-
-In addition each input or output device may support so-called private
-options, which are specific for that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the device
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-@c man end DEVICE OPTIONS
-
-@include indevs.texi
-@include outdevs.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavdevice.html,libavdevice}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-devices
-@settitle FFmpeg devices
-
-@end ignore
-
-@bye

doc/ffmpeg-filters.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Filters Documentation
-@titlepage
-@center @titlefont{FFmpeg Filters Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes filters, sources, and sinks provided by the
-libavfilter library.
-
-@c man end DESCRIPTION
-
-@include filters.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavfilter.html,libavfilter}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-filters
-@settitle FFmpeg filters
-
-@end ignore
-
-@bye

doc/ffmpeg-formats.texi

@@ -1,177 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Formats Documentation
-@titlepage
-@center @titlefont{FFmpeg Formats Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the supported formats (muxers and demuxers)
-provided by the libavformat library.
-
-@c man end DESCRIPTION
-
-@chapter Format Options
-@c man begin FORMAT OPTIONS
-
-The libavformat library provides some generic global options, which
-can be set on all the muxers and demuxers. In addition each muxer or
-demuxer may support so-called private options, which are specific for
-that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-The list of supported options follows:
-
-@table @option
-@item avioflags @var{flags} (@emph{input/output})
-Possible values:
-@table @samp
-@item direct
-Reduce buffering.
-@end table
-
-@item probesize @var{integer} (@emph{input})
-Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will allow to detect more
-information in case it is dispersed into the stream, but will increase
-latency. Must be an integer not lesser than 32. It is 5000000 by default.
-
-@item packetsize @var{integer} (@emph{output})
-Set packet size.
-
-@item fflags @var{flags} (@emph{input/output})
-Set format flags.
-
-Possible values:
-@table @samp
-@item ignidx
-Ignore index.
-@item genpts
-Generate PTS.
-@item nofillin
-Do not fill in missing values that can be exactly calculated.
-@item noparse
-Disable AVParsers, this needs @code{+nofillin} too.
-@item igndts
-Ignore DTS.
-@item discardcorrupt
-Discard corrupted frames.
-@item sortdts
-Try to interleave output packets by DTS.
-@item keepside
-Do not merge side data.
-@item latm
-Enable RTP MP4A-LATM payload.
-@item nobuffer
-Reduce the latency introduced by optional buffering
-@end table
-
-@item analyzeduration @var{integer} (@emph{input})
-Specify how many microseconds are analyzed to estimate duration.
-
-@item cryptokey @var{hexadecimal string} (@emph{input})
-Set decryption key.
-
-@item indexmem @var{integer} (@emph{input})
-Set max memory used for timestamp index (per stream).
-
-@item rtbufsize @var{integer} (@emph{input})
-Set max memory used for buffering real-time frames.
-
-@item fdebug @var{flags} (@emph{input/output})
-Print specific debug info.
-
-Possible values:
-@table @samp
-@item ts
-@end table
-
-@item max_delay @var{integer} (@emph{input/output})
-Set maximum muxing or demuxing delay in microseconds.
-
-@item fpsprobesize @var{integer} (@emph{input})
-Set number of frames used to probe fps.
-
-@item audio_preload @var{integer} (@emph{output})
-Set microseconds by which audio packets should be interleaved earlier.
-
-@item chunk_duration @var{integer} (@emph{output})
-Set microseconds for each chunk.
-
-@item chunk_size @var{integer} (@emph{output})
-Set size in bytes for each chunk.
-
-@item err_detect, f_err_detect @var{flags} (@emph{input})
-Set error detection flags. @code{f_err_detect} is deprecated and
-should be used only via the @command{ffmpeg} tool.
-
-Possible values:
-@table @samp
-@item crccheck
-Verify embedded CRCs.
-@item bitstream
-Detect bitstream specification deviations.
-@item buffer
-Detect improper bitstream length.
-@item explode
-Abort decoding on minor error detection.
-@item careful
-Consider things that violate the spec and have not been seen in the
-wild as errors.
-@item compliant
-Consider all spec non compliancies as errors.
-@item aggressive
-Consider things that a sane encoder should not do as an error.
-@end table
-
-@item use_wallclock_as_timestamps @var{integer} (@emph{input})
-Use wallclock as timestamps.
-
-@item avoid_negative_ts @var{integer} (@emph{output})
-Shift timestamps to make them positive. 1 enables, 0 disables, default
-of -1 enables when required by target format.
-
-@item skip_initial_bytes @var{integer} (@emph{input})
-Set number initial bytes to skip. Default is 0.
-
-@item correct_ts_overflow @var{integer} (@emph{input})
-Correct single timestamp overflows if set to 1. Default is 1.
-@end table
-
-@c man end FORMAT OPTIONS
-
-@include demuxers.texi
-@include muxers.texi
-@include metadata.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-formats
-@settitle FFmpeg formats
-
-@end ignore
-
-@bye

doc/ffmpeg-protocols.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Protocols Documentation
-@titlepage
-@center @titlefont{FFmpeg Protocols Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the input and output protocols provided by the
-libavformat library.
-
-@c man end DESCRIPTION
-
-@include protocols.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-protocols
-@settitle FFmpeg protocols
-
-@end ignore
-
-@bye

doc/ffmpeg-resampler.texi

@@ -1,238 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Resampler Documentation
-@titlepage
-@center @titlefont{FFmpeg Resampler Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The FFmpeg resampler provides an high-level interface to the
-libswresample library audio resampling utilities. In particular it
-allows to perform audio resampling, audio channel layout rematrixing,
-and convert audio format and packing layout.
-
-@c man end DESCRIPTION
-
-@chapter Resampler Options
-@c man begin RESAMPLER OPTIONS
-
-The audio resampler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, @var{option}=@var{value} for the aresample filter,
-by setting the value explicitly in the
-@code{SwrContext} options or using the @file{libavutil/opt.h} API for
-programmatic use.
-
-@table @option
-
-@item ich, in_channel_count
-Set the number of input channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{in_channel_layout} is set.
-
-@item och, out_channel_count
-Set the number of output channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{out_channel_layout} is set.
-
-@item uch, used_channel_count
-Set the number of used channels. Default value is 0. This option is
-only used for special remapping.
-
-@item isr, in_sample_rate
-Set the input sample rate. Default value is 0.
-
-@item osr, out_sample_rate
-Set the output sample rate. Default value is 0.
-
-@item isf, in_sample_fmt
-Specify the input sample format. It is set by default to @code{none}.
-
-@item osf, out_sample_fmt
-Specify the output sample format. It is set by default to @code{none}.
-
-@item tsf, internal_sample_fmt
-Set the internal sample format. Default value is @code{none}.
-
-@item icl, in_channel_layout
-Set the input channel layout.
-
-@item ocl, out_channel_layout
-Set the output channel layout.
-
-@item clev, center_mix_level
-Set center mix level. It is a value expressed in deciBel, and must be
-inclusively included between -32 and +32.
-
-@item slev, surround_mix_level
-Set surround mix level. It is a value expressed in deciBel, and must
-be inclusively included between -32 and +32.
-
-@item lfe_mix_evel
-Set LFE mix level.
-
-@item rmvol, rematrix_volume
-Set rematrix volume. Default value is 1.0.
-
-@item flags, swr_flags
-Set flags used by the converter. Default value is 0.
-
-It supports the following individual flags:
-@table @option
-@item res
-force resampling
-@end table
-
-@item dither_scale
-Set the dither scale. Default value is 1.
-
-@item dither_method
-Set dither method. Default value is 0.
-
-Supported values:
-@table @samp
-@item rectangular
-select rectangular dither
-@item triangular
-select triangular dither
-@item triangular_hp
-select triangular dither with high pass
-@end table
-
-@item resampler
-Set resampling engine. Default value is swr.
-
-Supported values:
-@table @samp
-@item swr
-select the native SW Resampler; filter options precision and cheby are not
-applicable in this case.
-@item soxr
-select the SoX Resampler (where available); compensation, and filter options
-filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
-case.
-@end table
-
-@item filter_size
-For swr only, set resampling filter size, default value is 32.
-
-@item phase_shift
-For swr only, set resampling phase shift, default value is 10, must be included
-between 0 and 30.
-
-@item linear_interp
-Use Linear Interpolation if set to 1, default value is 0.
-
-@item cutoff
-Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
-value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
-(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
-
-@item precision
-For soxr only, the precision in bits to which the resampled signal will be
-calculated.  The default value of 20 (which, with suitable dithering, is
-appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
-value of 28 gives SoX's 'Very High Quality'.
-
-@item cheby
-For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
-approximation for 'irrational' ratios. Default value is 0.
-
-@item async
-For swr only, simple 1 parameter audio sync to timestamps using stretching,
-squeezing, filling and trimming. Setting this to 1 will enable filling and
-trimming, larger values represent the maximum amount in samples that the data
-may be stretched or squeezed for each second.
-Default value is 0, thus no compensation is applied to make the samples match
-the audio timestamps.
-
-@item min_comp
-For swr only, set the minimum difference between timestamps and audio data (in
-seconds) to trigger stretching/squeezing/filling or trimming of the
-data to make it match the timestamps. The default is that
-stretching/squeezing/filling and trimming is disabled
-(@option{min_comp} = @code{FLT_MAX}).
-
-@item min_hard_comp
-For swr only, set the minimum difference between timestamps and audio data (in
-seconds) to trigger adding/dropping samples to make it match the
-timestamps.  This option effectively is a threshold to select between
-hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
-all compensation is by default disabled through @option{min_comp}.
-The default is 0.1.
-
-@item comp_duration
-For swr only, set duration (in seconds) over which data is stretched/squeezed
-to make it match the timestamps. Must be a non-negative double float value,
-default value is 1.0.
-
-@item max_soft_comp
-For swr only, set maximum factor by which data is stretched/squeezed to make it
-match the timestamps. Must be a non-negative double float value, default value
-is 0.
-
-@item matrix_encoding
-Select matrixed stereo encoding.
-
-It accepts the following values:
-@table @samp
-@item none
-select none
-@item dolby
-select Dolby
-@item dplii
-select Dolby Pro Logic II
-@end table
-
-Default value is @code{none}.
-
-@item filter_type
-For swr only, select resampling filter type. This only affects resampling
-operations.
-
-It accepts the following values:
-@table @samp
-@item cubic
-select cubic
-@item blackman_nuttall
-select Blackman Nuttall Windowed Sinc
-@item kaiser
-select Kaiser Windowed Sinc
-@end table
-
-@item kaiser_beta
-For swr only, set Kaiser Window Beta value. Must be an integer included between
-2 and 16, default value is 9.
-
-@end table
-
-@c man end RESAMPLER OPTIONS
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libswresample.html,libswresample}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-resampler
-@settitle FFmpeg Resampler
-
-@end ignore
-
-@bye

doc/ffmpeg-scaler.texi

@@ -1,141 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Scaler Documentation
-@titlepage
-@center @titlefont{FFmpeg Scaler Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The FFmpeg rescaler provides an high-level interface to the libswscale
-library image conversion utilities. In particular it allows to perform
-image rescaling and pixel format conversion.
-
-@c man end DESCRIPTION
-
-@chapter Scaler Options
-@c man begin SCALER OPTIONS
-
-The video scaler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools. For programmatic use, they can be set explicitly in the
-@code{SwsContext} options or through the @file{libavutil/opt.h} API.
-
-@table @option
-
-@item sws_flags
-Set the scaler flags. This is also used to set the scaling
-algorithm. Only a single algorithm should be selected.
-
-It accepts the following values:
-@table @samp
-@item fast_bilinear
-Select fast bilinear scaling algorithm.
-
-@item bilinear
-Select bilinear scaling algorithm.
-
-@item bicubic
-Select bicubic scaling algorithm.
-
-@item experimental
-Select experimental scaling algorithm.
-
-@item neighbor
-Select nearest neighbor rescaling algorithm.
-
-@item area
-Select averaging area rescaling algorithm.
-
-@item bicubiclin
-Select bicubic scaling algorithm for the luma component, bilinear for
-chroma components.
-
-@item gauss
-Select Gaussian rescaling algorithm.
-
-@item sinc
-Select sinc rescaling algorithm.
-
-@item lanczos
-Select lanczos rescaling algorithm.
-
-@item spline
-Select natural bicubic spline rescaling algorithm.
-
-@item print_info
-Enable printing/debug logging.
-
-@item accurate_rnd
-Enable accurate rounding.
-
-@item full_chroma_int
-Enable full chroma interpolation.
-
-@item full_chroma_inp
-Select full chroma input.
-
-@item bitexact
-Enable bitexact output.
-@end table
-
-@item srcw
-Set source width.
-
-@item srch
-Set source height.
-
-@item dstw
-Set destination width.
-
-@item dsth
-Set destination height.
-
-@item src_format
-Set source pixel format (must be expressed as an integer).
-
-@item dst_format
-Set destination pixel format (must be expressed as an integer).
-
-@item src_range
-Select source range.
-
-@item dst_range
-Select destination range.
-
-@item param0, param1
-Set scaling algorithm parameters. The specified values are specific of
-some scaling algorithms and ignored by others. The specified values
-are floating point number values.
-
-@end table
-
-@c man end SCALER OPTIONS
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libswscale.html,libswscale}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-scaler
-@settitle FFmpeg video scaling and pixel format converter
-
-@end ignore
-
-@bye

doc/ffmpeg-utils.texi

@@ -1,43 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Utilities Documentation
-@titlepage
-@center @titlefont{FFmpeg Utilities Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes some generic features and utilities provided
-by the libavutil library.
-
-@c man end DESCRIPTION
-
-@include syntax.texi
-@include eval.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-utils
-@settitle FFmpeg utilities
-
-@end ignore
-
-@bye

doc/ffmpeg.txt

@@ -29,7 +29,7 @@
                                                                           \                        /                                   :
                                                                     +======\======================/======+      ^                      :
                                                           ------> 0 |   : source_index  : st-:---        |      :                      :
-                    OutputFile output_files[]           /           +------------------------------------+      :                      :
+                    OuputFile output_files[]            /           +------------------------------------+      :                      :
                                                        /          1 |   :               :    :           |      :                      :
                  ^  +------+------------+-----+       /             +------------------------------------+      :                      :
                  :  |      : ost_index -:-----:------/            2 |   :               :    :           |      :                      :

doc/ffplay.texi

@@ -11,7 +11,11 @@
 
 @chapter Synopsis
 
-ffplay [@var{options}] [@file{input_file}]
+@example
+@c man begin SYNOPSIS
+ffplay [options] [@file{input_file}]
+@c man end
+@end example
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -130,20 +134,8 @@ Exit when video is done playing.
 Exit if any key is pressed.
 @item -exitonmousedown
 Exit if any mouse button is pressed.
-
-@item -codec:@var{media_specifier} @var{codec_name}
-Force a specific decoder implementation for the stream identified by
-@var{media_specifier}, which can assume the values @code{a} (audio),
-@code{v} (video), and @code{s} subtitle.
-
-@item -acodec @var{codec_name}
-Force a specific audio decoder.
-
-@item -vcodec @var{codec_name}
-Force a specific video decoder.
-
-@item -scodec @var{codec_name}
-Force a specific subtitle decoder.
+@item -codec:@var{stream_type}
+Force a specific decoder implementation
 @end table
 
 @section While playing
@@ -186,35 +178,28 @@ Seek to percentage in file corresponding to fraction of width.
 
 @c man end
 
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffprobe(1), ffserver(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
+@include eval.texi
+@include decoders.texi
+@include demuxers.texi
+@include muxers.texi
+@include indevs.texi
+@include outdevs.texi
+@include protocols.texi
+@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

doc/ffprobe.texi

@@ -11,7 +11,13 @@
 
 @chapter Synopsis
 
-ffprobe [@var{options}] [@file{input_file}]
+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
@@ -39,10 +45,6 @@ ffprobe output is designed to be easily parsable by a textual filter,
 and consists of one or more sections of a form defined by the selected
 writer, which is specified by the @option{print_format} option.
 
-Sections may contain other nested sections, and are identified by a
-name (which may be shared by other sections), and an unique
-name. See the output of @option{sections}.
-
 Metadata tags stored in the container or in the streams are recognized
 and printed in the corresponding "FORMAT" or "STREAM" section.
 
@@ -78,7 +80,7 @@ Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
 Prettify the format of the displayed values, it corresponds to the
 options "-unit -prefix -byte_binary_prefix -sexagesimal".
 
-@item -of, -print_format @var{writer_name}[=@var{writer_options}]
+@item -print_format @var{writer_name}[=@var{writer_options}]
 Set the output printing format.
 
 @var{writer_name} specifies the name of the writer, and
@@ -92,32 +94,6 @@ For example for printing the output in JSON format, specify:
 For more details on the available output printing formats, see the
 Writers section below.
 
-@item -sections
-Print sections structure and section information, and exit. The output
-is not meant to be parsed by a machine.
-
-@item -select_streams @var{stream_specifier}
-Select only the streams specified by @var{stream_specifier}. This
-option affects only the options related to streams
-(e.g. @code{show_streams}, @code{show_packets}, etc.).
-
-For example to show only audio streams, you can use the command:
-@example
-ffprobe -show_streams -select_streams a INPUT
-@end example
-
-To show only video packets belonging to the video stream with index 1:
-@example
-ffprobe -show_packets -select_streams v:1 INPUT
-@end example
-
-@item -show_data
-Show payload data, as an hexadecimal and ASCII dump. Coupled with
-@option{-show_packets}, it will dump the packets' data. Coupled with
-@option{-show_streams}, it will dump the codec extradata.
-
-The dump is printed as the "data" field. It may contain newlines.
-
 @item -show_error
 Show information about the error found when trying to probe the input.
 
@@ -130,64 +106,6 @@ stream.
 All the container format information is printed within a section with
 name "FORMAT".
 
-@item -show_format_entry @var{name}
-Like @option{-show_format}, but only prints the specified entry of the
-container format information, rather than all. This option may be given more
-than once, then all specified entries will be shown.
-
-This option is deprecated, use @code{show_entries} instead.
-
-@item -show_entries @var{section_entries}
-Set list of entries to show.
-
-Entries are specified according to the following
-syntax. @var{section_entries} contains a list of section entries
-separated by @code{:}. Each section entry is composed by a section
-name (or unique name), optionally followed by a list of entries local
-to that section, separated by @code{,}.
-
-If section name is specified but is followed by no @code{=}, all
-entries are printed to output, together with all the contained
-sections. Otherwise only the entries specified in the local section
-entries list are printed. In particular, if @code{=} is specified but
-the list of local entries is empty, then no entries will be shown for
-that section.
-
-Note that the order of specification of the local section entries is
-not honored in the output, and the usual display order will be
-retained.
-
-The formal syntax is given by:
-@example
-@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}]
-@var{SECTION_ENTRY}         ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]]
-@var{SECTION_ENTRIES}       ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}]
-@end example
-
-For example, to show only the index and type of each stream, and the PTS
-time, duration time, and stream index of the packets, you can specify
-the argument:
-@example
-packet=pts_time,duration_time,stream_index : stream=index,codec_type
-@end example
-
-To show all the entries in the section "format", but only the codec
-type in the section "stream", specify the argument:
-@example
-format : stream=codec_type
-@end example
-
-To show all the tags in the stream and format sections:
-@example
-format_tags : format_tags
-@end example
-
-To show only the @code{title} tag (if available) in the stream
-sections:
-@example
-stream_tags=title
-@end example
-
 @item -show_packets
 Show information about each packet contained in the input multimedia
 stream.
@@ -209,14 +127,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".
 
-@item -count_frames
-Count the number of frames per stream and report it in the
-corresponding stream section.
-
-@item -count_packets
-Count the number of packets per stream and report it in the
-corresponding stream section.
-
 @item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -240,10 +150,6 @@ Show information related to program and library versions. This is the
 equivalent of setting both @option{-show_program_version} and
 @option{-show_library_versions} options.
 
-@item -bitexact
-Force bitexact output, useful to produce output which is not dependent
-on the specific build.
-
 @item -i @var{input_file}
 Read @var{input_file}.
 
@@ -256,9 +162,8 @@ Read @var{input_file}.
 A writer defines the output format adopted by @command{ffprobe}, and will be
 used for printing all the parts of the output.
 
-A writer may accept one or more arguments, which specify the options
-to adopt. The options are specified as a list of @var{key}=@var{value}
-pairs, separated by ":".
+A writer may accept one or more arguments, which specify the options to
+adopt.
 
 A description of the currently available writers follows.
 
@@ -277,24 +182,8 @@ keyN=valN
 Metadata tags are printed as a line in the corresponding FORMAT or
 STREAM section, and are prefixed by the string "TAG:".
 
-A description of the accepted options follows.
-
-@table @option
-
-@item nokey, nk
-If set to 1 specify not to print the key of each field. Default value
-is 0.
-
-@item noprint_wrappers, nw
-If set to 1 specify not to print the section header and footer.
-Default value is 0.
-@end table
-
-@section compact, csv
-Compact and CSV format.
-
-The @code{csv} writer is equivalent to @code{compact}, but supports
-different defaults.
+@section compact
+Compact format.
 
 Each section is printed on a single line.
 If no option is specifid, the output has the form:
@@ -306,29 +195,30 @@ Metadata tags are printed in the corresponding "format" or "stream"
 section. A metadata tag key, if printed, is prefixed by the string
 "tag:".
 
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.
 
 @table @option
 
 @item item_sep, s
 Specify the character to use for separating fields in the output line.
-It must be a single printable character, it is "|" by default ("," for
-the @code{csv} writer).
+It must be a single printable character, it is "|" by default.
 
 @item nokey, nk
 If set to 1 specify not to print the key of each field. Its default
-value is 0 (1 for the @code{csv} writer).
+value is 0.
 
 @item escape, e
-Set the escape mode to use, default to "c" ("csv" for the @code{csv}
-writer).
+Set the escape mode to use, default to "c".
 
 It can assume one of the following values:
 @table @option
 @item c
-Perform C-like escaping. Strings containing a newline ('\n'), carriage
-return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
-character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
+Perform C-like escaping. Strings containing a newline ('\n') or
+carriage return ('\r'), the escaping character ('\') or the item
+separator character @var{SEP} are escaped using C-like fashioned
 escaping, so that a newline is converted to the sequence "\n", a
 carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
 converted to "\@var{SEP}".
@@ -342,83 +232,22 @@ containing a newline ('\n'), a carriage return ('\r'), a double quote
 Perform no escaping.
 @end table
 
-@item print_section, p
-Print the section name at the begin of each line if the value is
-@code{1}, disable it with value set to @code{0}. Default value is
-@code{1}.
-
 @end table
 
-@section flat
-Flat format.
+@section csv
+CSV format.
 
-A free-form output where each line contains an explicit key=value, such as
-"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be
-directly embedded in sh scripts as long as the separator character is an
-alphanumeric character or an underscore (see @var{sep_char} option).
-
-The description of the accepted options follows.
-
-@table @option
-@item sep_char, s
-Separator character used to separate the chapter, the section name, IDs and
-potential tags in the printed field key.
-
-Default value is '.'.
-
-@item hierarchical, h
-Specify if the section name specification should be hierarchical. If
-set to 1, and if there is more than one section in the current
-chapter, the section name will be prefixed by the name of the
-chapter. A value of 0 will disable this behavior.
-
-Default value is 1.
-@end table
-
-@section ini
-INI format output.
-
-Print output in an INI based format.
-
-The following conventions are adopted:
-
-@itemize
-@item
-all key and values are UTF-8
-@item
-'.' is the subgroup separator
-@item
-newline, '\t', '\f', '\b' and the following characters are escaped
-@item
-'\' is the escape character
-@item
-'#' is the comment indicator
-@item
-'=' is the key/value separator
-@item
-':' is not used but usually parsed as key/value separator
-@end itemize
-
-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-The description of the accepted options follows.
-
-@table @option
-@item hierarchical, h
-Specify if the section name specification should be hierarchical. If
-set to 1, and if there is more than one section in the current
-chapter, the section name will be prefixed by the name of the
-chapter. A value of 0 will disable this behavior.
-
-Default value is 1.
-@end table
+This writer is equivalent to
+@code{compact=item_sep=,:nokey=1:escape=csv}.
 
 @section json
 JSON based format.
 
 Each section is printed using JSON notation.
 
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.
 
 @table @option
@@ -436,15 +265,14 @@ XML based format.
 The XML output is described in the XML schema description file
 @file{ffprobe.xsd} installed in the FFmpeg datadir.
 
-An updated version of the schema can be retrieved at the url
-@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
-latest schema committed into the FFmpeg development source code tree.
-
 Note that the output issued will be compliant to the
 @file{ffprobe.xsd} schema only when no special global output options
 (@option{unit}, @option{prefix}, @option{byte_binary_prefix},
 @option{sexagesimal} etc.) are specified.
 
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.
 
 @table @option
@@ -463,59 +291,44 @@ This option automatically sets @option{fully_qualified} to 1.
 
 For more information about the XML format, see
 @url{http://www.w3.org/XML/}.
-@c man end WRITERS
 
 @chapter Timecode
-@c man begin TIMECODE
 
 @command{ffprobe} supports Timecode extraction:
 
 @itemize
 
-@item
-MPEG1/2 timecode is extracted from the GOP, and is available in the video
+@item MPEG1/2 timecode is extracted from the GOP, and is available in the video
 stream details (@option{-show_streams}, see @var{timecode}).
 
-@item
-MOV timecode is extracted from tmcd track, so is available in the tmcd
+@item MOV timecode is extracted from tmcd track, so is available in the tmcd
 stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
 
-@item
-DV, GXF and AVI timecodes are available in format metadata
+@item DV and GXF timecodes are available in format metadata
 (@option{-show_format}, see @var{TAG:timecode}).
 
 @end itemize
-@c man end TIMECODE
 
-@chapter See Also
+@c man end WRITERS
 
-@ifhtml
-@url{ffplay.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffserver(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
+@include decoders.texi
+@include demuxers.texi
+@include protocols.texi
+@include indevs.texi
 
 @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

doc/ffprobe.xsd

@@ -39,12 +39,9 @@
       <xsd:attribute name="dts_time"      type="xsd:float" />
       <xsd:attribute name="duration"      type="xsd:long"  />
       <xsd:attribute name="duration_time" type="xsd:float" />
-      <xsd:attribute name="convergence_duration"      type="xsd:long"  />
-      <xsd:attribute name="convergence_duration_time" type="xsd:float" />
       <xsd:attribute name="size"          type="xsd:long" use="required" />
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
-      <xsd:attribute name="data"          type="xsd:string" />
     </xsd:complexType>
 
     <xsd:complexType name="frameType">
@@ -56,16 +53,11 @@
       <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="pkt_duration"  type="xsd:long" />
-      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
       <xsd:attribute name="pkt_pos"       type="xsd:long" />
-      <xsd:attribute name="pkt_size"      type="xsd:int" />
 
       <!-- audio attributes -->
       <xsd:attribute name="sample_fmt"             type="xsd:string"/>
       <xsd:attribute name="nb_samples"             type="xsd:long"  />
-      <xsd:attribute name="channels"               type="xsd:int"   />
-      <xsd:attribute name="channel_layout"         type="xsd:string"/>
 
       <!-- video attributes -->
       <xsd:attribute name="width"                  type="xsd:long"  />
@@ -87,35 +79,14 @@
         </xsd:sequence>
     </xsd:complexType>
 
-    <xsd:complexType name="streamDispositionType">
-      <xsd:attribute name="default"          type="xsd:int" use="required" />
-      <xsd:attribute name="dub"              type="xsd:int" use="required" />
-      <xsd:attribute name="original"         type="xsd:int" use="required" />
-      <xsd:attribute name="comment"          type="xsd:int" use="required" />
-      <xsd:attribute name="lyrics"           type="xsd:int" use="required" />
-      <xsd:attribute name="karaoke"          type="xsd:int" use="required" />
-      <xsd:attribute name="forced"           type="xsd:int" use="required" />
-      <xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
-      <xsd:attribute name="visual_impaired"  type="xsd:int" use="required" />
-      <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
-      <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
-    </xsd:complexType>
-
     <xsd:complexType name="streamType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
       <xsd:attribute name="index"            type="xsd:int" use="required"/>
       <xsd:attribute name="codec_name"       type="xsd:string" />
       <xsd:attribute name="codec_long_name"  type="xsd:string" />
-      <xsd:attribute name="profile"          type="xsd:string" />
       <xsd:attribute name="codec_type"       type="xsd:string" />
       <xsd:attribute name="codec_time_base"  type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
-      <xsd:attribute name="extradata"        type="xsd:string" />
 
       <!-- video attributes -->
       <xsd:attribute name="width"                type="xsd:int"/>
@@ -137,14 +108,9 @@
       <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
       <xsd:attribute name="avg_frame_rate"   type="xsd:string" use="required"/>
       <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
-      <xsd:attribute name="start_pts"        type="xsd:long"/>
       <xsd:attribute name="start_time"       type="xsd:float"/>
-      <xsd:attribute name="duration_ts"      type="xsd:long"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
-      <xsd:attribute name="bit_rate"         type="xsd:int"/>
       <xsd:attribute name="nb_frames"        type="xsd:int"/>
-      <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
-      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
     </xsd:complexType>
 
     <xsd:complexType name="formatType">
@@ -155,7 +121,7 @@
       <xsd:attribute name="filename"         type="xsd:string" use="required"/>
       <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
       <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
-      <xsd:attribute name="format_long_name" type="xsd:string"/>
+      <xsd:attribute name="format_long_name" type="xsd:string" use="required"/>
       <xsd:attribute name="start_time"       type="xsd:float"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="size"             type="xsd:long"/>
@@ -188,7 +154,6 @@
       <xsd:attribute name="minor"       type="xsd:int"    use="required"/>
       <xsd:attribute name="micro"       type="xsd:int"    use="required"/>
       <xsd:attribute name="version"     type="xsd:int"    use="required"/>
-      <xsd:attribute name="ident"       type="xsd:string" use="required"/>
     </xsd:complexType>
 
     <xsd:complexType name="libraryVersionsType">

doc/ffserver.conf

@@ -25,6 +25,10 @@ MaxBandwidth 1000
 # '-' 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
@@ -369,3 +373,5 @@ ACL allow 192.168.0.0 192.168.255.255
 <Redirect index.html>
 URL http://www.ffmpeg.org/
 </Redirect>
+
+

doc/ffserver.texi

@@ -9,9 +9,15 @@
 
 @contents
 
-@chapter Synopsis
+@chapter Synopsys
 
-ffserver [@var{options}]
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffserver [options]
+@c man end
+@end example
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -22,6 +28,11 @@ 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.html} for more
@@ -81,7 +92,7 @@ to make it work correctly.
 
 @section What do I need?
 
-I use Linux on a 900 MHz Duron with a cheap Bt848 based TV capture card. I'm
+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.]
 
@@ -227,19 +238,6 @@ 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
 
-@section What is FFM, FFM2
-
-FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
-video and audio streams and encoding options, and can store a moving time segment
-of an infinite movie or a whole movie.
-
-FFM is version specific, and there is limited compatibility of FFM files
-generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but its not guaranteed to work.
-
-FFM2 is extensible while maintaining compatibility and should work between
-differing versions of tools. FFM2 is the default.
-
 @chapter Options
 @c man begin OPTIONS
 
@@ -256,40 +254,26 @@ within the various <Stream> sections. Since ffserver will not launch
 any ffmpeg instances, you will have to launch them manually.
 @item -d
 Enable debug mode. This option increases log verbosity, directs log
-messages to stdout.
+messages to stdout and causes ffserver to run in the foreground
+rather than as a daemon.
 @end table
 @c man end
 
-@chapter See Also
-
-@ifhtml
-The @file{doc/ffserver.conf} example,
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-The @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
-
 @ignore
 
 @setfilename ffserver
 @settitle ffserver video server
 
+@c man begin SEEALSO
+
+ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffserver.conf}
+example and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
 @end ignore
 
 @bye

doc/filter_design.txt

@@ -1,269 +0,0 @@
-Filter design
-=============
-
-This document explains guidelines that should be observed (or ignored with
-good reason) when writing filters for libavfilter.
-
-In this document, the word “frame” indicates either a video frame or a group
-of audio samples, as stored in an AVFilterBuffer structure.
-
-
-Format negotiation
-==================
-
-  The query_formats method should set, for each input and each output links,
-  the list of supported formats.
-
-  For video links, that means pixel format. For audio links, that means
-  channel layout, and sample format (the sample packing is implied by the
-  sample format).
-
-  The lists are not just lists, they are references to shared objects. When
-  the negotiation mechanism computes the intersection of the formats
-  supported at each ends of a link, all references to both lists are
-  replaced with a reference to the intersection. And when a single format is
-  eventually chosen for a link amongst the remaining list, again, all
-  references to the list are updated.
-
-  That means that if a filter requires that its input and output have the
-  same format amongst a supported list, all it has to do is use a reference
-  to the same list of formats.
-
-
-Buffer references ownership and permissions
-===========================================
-
-  Principle
-  ---------
-
-    Audio and video data are voluminous; the buffer and buffer reference
-    mechanism is intended to avoid, as much as possible, expensive copies of
-    that data while still allowing the filters to produce correct results.
-
-    The data is stored in buffers represented by AVFilterBuffer structures.
-    They must not be accessed directly, but through references stored in
-    AVFilterBufferRef structures. Several references can point to the
-    same buffer; the buffer is automatically deallocated once all
-    corresponding references have been destroyed.
-
-    The characteristics of the data (resolution, sample rate, etc.) are
-    stored in the reference; different references for the same buffer can
-    show different characteristics. In particular, a video reference can
-    point to only a part of a video buffer.
-
-    A reference is usually obtained as input to the start_frame or
-    filter_frame method or requested using the ff_get_video_buffer or
-    ff_get_audio_buffer functions. A new reference on an existing buffer can
-    be created with the avfilter_ref_buffer. A reference is destroyed using
-    the avfilter_unref_bufferp function.
-
-  Reference ownership
-  -------------------
-
-    At any time, a reference “belongs” to a particular piece of code,
-    usually a filter. With a few caveats that will be explained below, only
-    that piece of code is allowed to access it. It is also responsible for
-    destroying it, although this is sometimes done automatically (see the
-    section on link reference fields).
-
-    Here are the (fairly obvious) rules for reference ownership:
-
-    * A reference received by the start_frame or filter_frame method
-      belong to the corresponding filter.
-
-      Special exception: for video references: the reference may be used
-      internally for automatic copying and must not be destroyed before
-      end_frame; it can be given away to ff_start_frame.
-
-    * A reference passed to ff_start_frame or ff_filter_frame is given
-      away and must no longer be used.
-
-    * A reference created with avfilter_ref_buffer belongs to the code that
-      created it.
-
-    * A reference obtained with ff_get_video_buffer or ff_get_audio_buffer
-      belongs to the code that requested it.
-
-    * A reference given as return value by the get_video_buffer or
-      get_audio_buffer method is given away and must no longer be used.
-
-  Link reference fields
-  ---------------------
-
-    The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
-    the rules to handle them:
-
-    * cur_buf is set before the start_frame and filter_frame methods to
-      the same reference given as argument to the methods and belongs to the
-      destination filter of the link. If it has not been cleared after
-      end_frame or filter_frame, libavfilter will automatically destroy
-      the reference; therefore, any filter that needs to keep the reference
-      for longer must set cur_buf to NULL.
-
-    * out_buf belongs to the source filter of the link and can be used to
-      store a reference to the buffer that has been sent to the destination.
-      If it is not NULL after end_frame or filter_frame, libavfilter will
-      automatically destroy the reference.
-
-      If a video input pad does not have a start_frame method, the default
-      method will request a buffer on the first output of the filter, store
-      the reference in out_buf and push a second reference to the output.
-
-    * src_buf, cur_buf_copy and partial_buf are used by libavfilter
-      internally and must not be accessed by filters.
-
-  Reference permissions
-  ---------------------
-
-    The AVFilterBufferRef structure has a perms field that describes what
-    the code that owns the reference is allowed to do to the buffer data.
-    Different references for the same buffer can have different permissions.
-
-    For video filters, the permissions only apply to the parts of the buffer
-    that have already been covered by the draw_slice method.
-
-    The value is a binary OR of the following constants:
-
-    * AV_PERM_READ: the owner can read the buffer data; this is essentially
-      always true and is there for self-documentation.
-
-    * AV_PERM_WRITE: the owner can modify the buffer data.
-
-    * AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data
-      will not be modified by previous filters.
-
-    * AV_PERM_REUSE: the owner can output the buffer several times, without
-      modifying the data in between.
-
-    * AV_PERM_REUSE2: the owner can output the buffer several times and
-      modify the data in between (useless without the WRITE permissions).
-
-    * AV_PERM_ALIGN: the owner can access the data using fast operations
-      that require data alignment.
-
-    The READ, WRITE and PRESERVE permissions are about sharing the same
-    buffer between several filters to avoid expensive copies without them
-    doing conflicting changes on the data.
-
-    The REUSE and REUSE2 permissions are about special memory for direct
-    rendering. For example a buffer directly allocated in video memory must
-    not modified once it is displayed on screen, or it will cause tearing;
-    it will therefore not have the REUSE2 permission.
-
-    The ALIGN permission is about extracting part of the buffer, for
-    copy-less padding or cropping for example.
-
-
-    References received on input pads are guaranteed to have all the
-    permissions stated in the min_perms field and none of the permissions
-    stated in the rej_perms.
-
-    References obtained by ff_get_video_buffer and ff_get_audio_buffer are
-    guaranteed to have at least all the permissions requested as argument.
-
-    References created by avfilter_ref_buffer have the same permissions as
-    the original reference minus the ones explicitly masked; the mask is
-    usually ~0 to keep the same permissions.
-
-    Filters should remove permissions on reference they give to output
-    whenever necessary. It can be automatically done by setting the
-    rej_perms field on the output pad.
-
-    Here are a few guidelines corresponding to common situations:
-
-    * Filters that modify and forward their frame (like drawtext) need the
-      WRITE permission.
-
-    * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and and must request a buffer
-      with the WRITE permission.
-
-    * Filters that intend to keep a reference after the filtering process
-      is finished (after end_frame or filter_frame returns) must have the
-      PRESERVE permission on it and remove the WRITE permission if they
-      create a new reference to give it away.
-
-    * Filters that intend to modify a reference they have kept after the end
-      of the filtering process need the REUSE2 permission and must remove
-      the PRESERVE permission if they create a new reference to give it
-      away.
-
-
-Frame scheduling
-================
-
-  The purpose of these rules is to ensure that frames flow in the filter
-  graph without getting stuck and accumulating somewhere.
-
-  Simple filters that output one frame for each input frame should not have
-  to worry about it.
-
-  start_frame / filter_frame
-  ----------------------------
-
-    These methods are called when a frame is pushed to the filter's input.
-    They can be called at any time except in a reentrant way.
-
-    If the input frame is enough to produce output, then the filter should
-    push the output frames on the output link immediately.
-
-    As an exception to the previous rule, if the input frame is enough to
-    produce several output frames, then the filter needs output only at
-    least one per link. The additional frames can be left buffered in the
-    filter; these buffered frames must be flushed immediately if a new input
-    produces new output.
-
-    (Example: framerate-doubling filter: start_frame must (1) flush the
-    second copy of the previous frame, if it is still there, (2) push the
-    first copy of the incoming frame, (3) keep the second copy for later.)
-
-    If the input frame is not enough to produce output, the filter must not
-    call request_frame to get more. It must just process the frame or queue
-    it. The task of requesting more frames is left to the filter's
-    request_frame method or the application.
-
-    If a filter has several inputs, the filter must be ready for frames
-    arriving randomly on any input. Therefore, any filter with several inputs
-    will most likely require some kind of queuing mechanism. It is perfectly
-    acceptable to have a limited queue and to drop frames when the inputs
-    are too unbalanced.
-
-  request_frame
-  -------------
-
-    This method is called when a frame is wanted on an output.
-
-    For an input, it should directly call start_frame or filter_frame on
-    the corresponding output.
-
-    For a filter, if there are queued frames already ready, one of these
-    frames should be pushed. If not, the filter should request a frame on
-    one of its inputs, repeatedly until at least one frame has been pushed.
-
-    Return values:
-    if request_frame could produce a frame, it should return 0;
-    if it could not for temporary reasons, it should return AVERROR(EAGAIN);
-    if it could not because there are no more frames, it should return
-    AVERROR_EOF.
-
-    The typical implementation of request_frame for a filter with several
-    inputs will look like that:
-
-        if (frames_queued) {
-            push_one_frame();
-            return 0;
-        }
-        while (!frame_pushed) {
-            input = input_where_a_frame_is_most_needed();
-            ret = avfilter_request_frame(input);
-            if (ret == AVERROR_EOF) {
-                process_eof_on_input();
-            } else if (ret < 0) {
-                return ret;
-            }
-        }
-        return 0;
-
-    Note that, except for filters that can have queued frames, request_frame
-    does not push frames: it requests them to its input, and as a reaction,
-    the start_frame / filter_frame method will be called and do the work.

doc/general.texi

@@ -13,8 +13,7 @@
 
 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
-@command{./configure}.
+explicitly requested by passing the appropriate flags to @file{./configure}.
 
 @section OpenJPEG
 
@@ -24,22 +23,17 @@ instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
 @file{./configure}.
 
 
-@section OpenCORE, VisualOn, and Fraunhofer libraries
+@section OpenCORE and VisualOn libraries
 
-Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
-libraries provide encoders for a number of audio codecs.
+Spun off Google Android sources, OpenCore and VisualOn libraries provide
+encoders for a number of audio codecs.
 
 @float NOTE
 OpenCORE and VisualOn libraries are under the Apache License 2.0
 (see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
-incompatible to the LGPL version 2.1 and GPL version 2. You have to
+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) by passing @code{--enable-version3} to configure in
-order to use it.
-
-The Fraunhofer AAC library is licensed under a license incompatible to the GPL
-and is not known to be compatible to the LGPL. Therefore, you have to pass
-@code{--enable-nonfree} to configure to use it.
+GPL components, GPL version 3) to use it.
 @end float
 
 @subsection OpenCORE AMR
@@ -68,14 +62,6 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
 
-@subsection Fraunhofer AAC library
-
-FFmpeg can make use of the Fraunhofer AAC library for AAC encoding.
-
-Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libfdk-aac} to configure to enable it.
-
 @section LAME
 
 FFmpeg can make use of the LAME library for MP3 encoding.
@@ -84,14 +70,6 @@ Go to @url{http://lame.sourceforge.net/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libmp3lame} to configure to enable it.
 
-@section TwoLAME
-
-FFmpeg can make use of the TwoLAME library for MP2 encoding.
-
-Go to @url{http://www.twolame.org/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libtwolame} to configure to enable it.
-
 @section libvpx
 
 FFmpeg can make use of the libvpx library for VP8 encoding.
@@ -114,17 +92,6 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section libilbc
-
-iLBC is a narrowband speech codec that has been made freely available
-by Google as part of the WebRTC project. libilbc is a packaging friendly
-copy of the iLBC codec. FFmpeg can make use of the libilbc library for
-iLBC encoding and decoding.
-
-Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
-installing the library. Then pass @code{--enable-libilbc} to configure to
-enable it.
-
 
 
 @chapter Supported File Formats, Codecs or Features
@@ -148,19 +115,11 @@ library:
 @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 Amazing Studio Packed Animation File        @tab   @tab X
-    @tab Multimedia format used in game Heart Of Darkness.
 @item Apple HTTP Live Streaming @tab   @tab X
 @item Artworx Data Format       @tab   @tab X
-@item AFC                       @tab   @tab X
-    @tab Audio format used on the Nintendo Gamecube.
 @item ASF                       @tab X @tab X
-@item AST                       @tab X @tab X
-    @tab Audio format used on the Nintendo Wii.
 @item AVI                       @tab X @tab X
 @item AVISynth                  @tab   @tab X
-@item AVR                       @tab   @tab X
-    @tab Audio format used on Mac.
 @item AVS                       @tab   @tab X
     @tab Multimedia format used by the Creature Shock game.
 @item Beam Software SIFF        @tab   @tab X
@@ -174,8 +133,6 @@ library:
     @tab Used in Z and Z95 games.
 @item Brute Force & Ignorance   @tab   @tab X
     @tab Used in the game Flash Traffic: City of Angels.
-@item BRSTM                     @tab   @tab X
-    @tab Audio format used on the Nintendo Wii.
 @item BWF                       @tab X @tab X
 @item CRI ADX                   @tab X @tab X
     @tab Audio-only format used in console video games.
@@ -186,8 +143,6 @@ library:
     @tab Multimedia format used by Delphine Software games.
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
-@item Commodore CDXL            @tab   @tab X
-    @tab Amiga CD video format
 @item Core Audio Format         @tab X @tab X
     @tab Apple Core Audio Format
 @item CRC testing format        @tab X @tab
@@ -206,7 +161,6 @@ library:
 @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 Ensoniq Paris Audio File  @tab   @tab X
 @item FFM (FFserver live feed)  @tab X @tab X
 @item Flash (SWF)               @tab X @tab X
 @item Flash 9 (AVM2)            @tab X @tab X
@@ -221,12 +175,12 @@ library:
 @item G.723.1                   @tab X @tab X
 @item G.729 BIT                 @tab X @tab X
 @item G.729 raw                 @tab   @tab X
-@item GIF Animation             @tab X @tab X
+@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 iCEDraw File              @tab   @tab X
-@item ICO                       @tab X @tab X
+@item ICO                       @tab   @tab X
     @tab Microsoft Windows ICO
 @item id Quake II CIN video     @tab   @tab X
 @item id RoQ                    @tab X @tab X
@@ -234,20 +188,17 @@ library:
 @item IEC61937 encapsulation @tab X @tab X
 @item IFF                       @tab   @tab X
     @tab Interchange File Format
-@item iLBC                      @tab X @tab X
 @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 X @tab X
     @tab A format used by libvpx
-@item IRCAM                     @tab X @tab X
 @item LATM                      @tab X @tab X
 @item LMLM4                     @tab   @tab X
     @tab Used by Linux Media Labs MPEG-4 PCI boards
 @item LOAS                      @tab   @tab X
     @tab contains LATM multiplexed AAC audio
-@item LVF                       @tab   @tab X
 @item LXF                       @tab   @tab X
     @tab VR native stream format, used by Leitch/Harris' video servers.
 @item Matroska                  @tab X @tab X
@@ -257,9 +208,6 @@ library:
 @item MAXIS XA                  @tab   @tab X
     @tab Used in Sim City 3000; file extension .xa.
 @item MD Studio                 @tab   @tab X
-@item Metal Gear Solid: The Twin Snakes @tab @tab X
-@item Megalux Frame             @tab   @tab X
-    @tab Used by Megalux Ultimate Paint
 @item Mobotix .mxg              @tab   @tab X
 @item Monkey's Audio            @tab   @tab X
 @item Motion Pixels MVI         @tab   @tab X
@@ -287,7 +235,6 @@ library:
     @tab SMPTE 386M, D-10/IMX Mapping.
 @item NC camera feed            @tab   @tab X
     @tab NC (AVIP NC4600) camera streams
-@item NIST SPeech HEader REsources @tab   @tab X
 @item NTT TwinVQ (VQF)          @tab   @tab X
     @tab Nippon Telegraph and Telephone Corporation TwinVQ.
 @item Nullsoft Streaming Video  @tab   @tab X
@@ -296,7 +243,6 @@ library:
     @tab NUT Open Container Format
 @item Ogg                       @tab X @tab X
 @item Playstation Portable PMP  @tab   @tab X
-@item Portable Voice Format     @tab   @tab X
 @item TechnoTrend PVA           @tab   @tab X
     @tab Used by TechnoTrend DVB PCI boards.
 @item QCP                       @tab   @tab X
@@ -307,7 +253,6 @@ library:
 @item raw Dirac                 @tab X @tab X
 @item raw DNxHD                 @tab X @tab X
 @item raw DTS                   @tab X @tab X
-@item raw DTS-HD                @tab   @tab X
 @item raw E-AC-3                @tab X @tab X
 @item raw FLAC                  @tab X @tab X
 @item raw GSM                   @tab   @tab X
@@ -325,7 +270,6 @@ library:
 @item raw video                 @tab X @tab X
 @item raw id RoQ                @tab X @tab
 @item raw Shorten               @tab   @tab X
-@item raw TAK                   @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
@@ -367,7 +311,6 @@ library:
 @item SDP                       @tab   @tab X
 @item Sega FILM/CPK             @tab   @tab X
     @tab Used in many Sega Saturn console games.
-@item Silicon Graphics Movie    @tab   @tab X
 @item Sierra SOL                @tab   @tab X
     @tab .sol files used in Sierra Online games.
 @item Sierra VMD                @tab   @tab X
@@ -376,12 +319,10 @@ library:
     @tab Multimedia format used by many games.
 @item SMJPEG                    @tab X @tab X
     @tab Used in certain Loki game ports.
-@item Smush                     @tab   @tab X
-    @tab Multimedia format used in some LucasArts games.
 @item Sony OpenMG (OMA)         @tab X @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 X @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 Text files                @tab   @tab X
@@ -391,9 +332,8 @@ library:
     @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 Vivo                      @tab   @tab X
 @item WAV                       @tab X @tab X
-@item WavPack                   @tab X @tab X
+@item WavPack                   @tab   @tab X
 @item WebM                      @tab X @tab X
 @item Windows Televison (WTV)   @tab X @tab X
 @item Wing Commander III movie  @tab   @tab X
@@ -426,12 +366,8 @@ following image formats are supported:
     @tab Only uncompressed GIFs are generated.
 @item BMP          @tab X @tab X
     @tab Microsoft BMP image
-@item PIX          @tab   @tab X
-    @tab PIX is an image format used in the Argonaut BRender engine.
 @item DPX          @tab X @tab X
     @tab Digital Picture Exchange
-@item EXR          @tab   @tab X
-    @tab OpenEXR
 @item JPEG         @tab X @tab X
     @tab Progressive JPEG is not supported.
 @item JPEG 2000    @tab X @tab X
@@ -457,16 +393,12 @@ following image formats are supported:
     @tab V.Flash PTX format
 @item SGI          @tab X @tab X
     @tab SGI RGB image format
-@item Sun Rasterfile  @tab X @tab X
+@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
-@item XBM  @tab X @tab X
-    @tab X BitMap image format
-@item XFace @tab X @tab X
-    @tab X-Face image format
 @item XWD  @tab X @tab X
     @tab X Window Dump image format
 @end multitable
@@ -482,9 +414,10 @@ following image formats are supported:
 @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 A64 multicolor         @tab  X  @tab
     @tab Creates video suitable to be played on a commodore 64 (multicolor mode).
-@item Amazing Studio PAF Video @tab     @tab  X
 @item American Laser Games MM  @tab    @tab X
     @tab Used in games like Mad Dog McCree.
 @item AMV Video              @tab  X  @tab  X
@@ -511,8 +444,6 @@ following image formats are supported:
     @tab fourcc: AVrp
 @item AVS (Audio Video Standard) video  @tab     @tab  X
     @tab Video encoding used by the Creature Shock game.
-@item AYUV                   @tab  X  @tab  X
-    @tab Microsoft uncompressed packed 4:4:4:4
 @item Beam Software VB       @tab     @tab  X
 @item Bethesda VID video     @tab     @tab  X
     @tab Used in some games from Bethesda Softworks.
@@ -527,23 +458,19 @@ following image formats are supported:
     @tab fourcc: CSCD
 @item CD+G                   @tab     @tab  X
     @tab Video codec for CD+G karaoke disks
-@item CDXL                   @tab     @tab  X
-    @tab Amiga CD video codec
 @item Chinese AVS video      @tab  E  @tab  X
     @tab AVS1-P2, JiZhun profile, encoding through external library libxavs
 @item Delphine Software International CIN video  @tab     @tab  X
     @tab Codec used in Delphine Software International games.
 @item Discworld II BMV Video @tab     @tab  X
-@item Canopus Lossless Codec @tab     @tab  X
 @item Cinepak                @tab     @tab  X
 @item Cirrus Logic AccuPak   @tab  X  @tab  X
     @tab fourcc: CLJR
-@item CPiA Video Format      @tab     @tab  X
 @item Creative YUV (CYUV)    @tab     @tab  X
 @item DFA                    @tab     @tab  X
     @tab Codec used in Chronomaster game.
 @item Dirac                  @tab  E  @tab  X
-    @tab supported through external library libschroedinger
+    @tab supported through external libdirac/libschroedinger libraries
 @item Deluxe Paint Animation @tab     @tab  X
 @item DNxHD                  @tab   X @tab  X
     @tab aka SMPTE VC3
@@ -564,13 +491,12 @@ following image formats are supported:
 @item Escape 124             @tab     @tab  X
 @item Escape 130             @tab     @tab  X
 @item FFmpeg video codec #1  @tab  X  @tab  X
-    @tab lossless codec (fourcc: FFV1)
+    @tab experimental lossless codec (fourcc: FFV1)
 @item Flash Screen Video v1  @tab  X  @tab  X
     @tab fourcc: FSV1
 @item Flash Screen Video v2  @tab  X  @tab  X
 @item Flash Video (FLV)      @tab  X  @tab  X
     @tab Sorenson H.263 used in Flash
-@item Forward Uncompressed   @tab     @tab  X
 @item Fraps                  @tab     @tab  X
 @item H.261                  @tab  X  @tab  X
 @item H.263 / H.263-1996     @tab  X  @tab  X
@@ -608,18 +534,8 @@ following image formats are supported:
 @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 LucasArts Smush        @tab     @tab  X
-    @tab Used in LucasArts games.
 @item lossless MJPEG         @tab  X  @tab  X
-@item Microsoft ATC Screen   @tab     @tab  X
-    @tab Also known as Microsoft Screen 3.
-@item Microsoft Expression Encoder Screen  @tab     @tab  X
-    @tab Also known as Microsoft Titanium Screen 2.
 @item Microsoft RLE          @tab     @tab  X
-@item Microsoft Screen 1     @tab     @tab  X
-    @tab Also known as Windows Media Video V7 Screen.
-@item Microsoft Screen 2     @tab     @tab  X
-    @tab Also known as Windows Media Video V9 Screen.
 @item Microsoft Video 1      @tab     @tab  X
 @item Mimic                  @tab     @tab  X
     @tab Used in MSN Messenger Webcam streams.
@@ -648,8 +564,8 @@ following image formats are supported:
     @tab fourcc: VP60,VP61,VP62
 @item VP8                    @tab  E  @tab  X
     @tab fourcc: VP80, encoding supported through external library libvpx
-@item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
-    @tab fourcc: Y216
+@item planar RGB             @tab     @tab  X
+    @tab fourcc: 8BPS
 @item Prores                 @tab     @tab  X
     @tab fourcc: apch,apcn,apcs,apco
 @item Q-team QPEG            @tab     @tab  X
@@ -673,11 +589,8 @@ following image formats are supported:
     @tab Texture dictionaries used by the Renderware Engine.
 @item RL2 video              @tab     @tab  X
     @tab used in some games by Entertainment Software Partners
-@item SGI RLE 8-bit          @tab     @tab  X
 @item Sierra VMD video       @tab     @tab  X
     @tab Used in Sierra VMD files.
-@item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
-@item Silicon Graphics Motion Video Compressor 2 (MVC2)  @tab     @tab  X
 @item Smacker video          @tab     @tab  X
     @tab Video encoding used in Smacker.
 @item SMPTE VC-1             @tab     @tab  X
@@ -692,16 +605,13 @@ following image formats are supported:
     @tab fourcc: SP5X
 @item TechSmith Screen Capture Codec  @tab     @tab  X
     @tab fourcc: TSCC
-@item TechSmith Screen Capture Codec 2  @tab     @tab  X
-    @tab fourcc: TSC2
 @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 Ut Video               @tab  X  @tab  X
+@item Ut Video               @tab     @tab  X
 @item v210 QuickTime uncompressed 4:2:2 10-bit     @tab  X  @tab  X
 @item v308 QuickTime uncompressed 4:4:4            @tab  X  @tab  X
-@item v408 QuickTime uncompressed 4:4:4:4          @tab  X  @tab  X
 @item v410 QuickTime uncompressed 4:4:4 10-bit     @tab  X  @tab  X
 @item VBLE Lossless Codec    @tab     @tab  X
 @item VMware Screen Codec / VMware Video  @tab     @tab  X
@@ -722,7 +632,6 @@ following image formats are supported:
 @item Psygnosis YOP Video    @tab     @tab  X
 @item yuv4                   @tab  X  @tab  X
     @tab libquicktime uncompressed packed 4:2:0
-@item ZeroCodec Lossless 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
@@ -737,8 +646,7 @@ following image formats are supported:
 
 @multitable @columnfractions .4 .1 .1 .4
 @item Name @tab Encoding @tab Decoding @tab Comments
-@item 8SVX exponential       @tab     @tab  X
-@item 8SVX fibonacci         @tab     @tab  X
+@item 8SVX audio             @tab     @tab  X
 @item AAC+                   @tab  E  @tab  X
     @tab encoding supported through external library libaacplus
 @item AAC                    @tab  E  @tab  X
@@ -769,19 +677,19 @@ following image formats are supported:
 @item ADPCM IMA Westwood     @tab     @tab  X
 @item ADPCM ISS IMA          @tab     @tab  X
     @tab Used in FunCom games.
-@item ADPCM IMA Dialogic     @tab     @tab  X
 @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 AFC  @tab     @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
@@ -792,7 +700,6 @@ following image formats are supported:
     @tab encoding supported through external library libopencore-amrnb
 @item AMR-WB                 @tab  E  @tab  X
     @tab encoding supported through external library libvo-amrwbenc
-@item Amazing Studio PAF Audio @tab     @tab  X
 @item Apple lossless audio   @tab  X  @tab  X
     @tab QuickTime fourcc 'alac'
 @item Atrac 1                @tab     @tab  X
@@ -826,9 +733,6 @@ following image formats are supported:
     @tab encoding supported through external library libgsm
 @item GSM Microsoft variant  @tab  E  @tab  X
     @tab encoding supported through external library libgsm
-@item IAC (Indeo Audio Coder)  @tab     @tab  X
-@item iLBC (Internet Low Bitrate Codec) @tab  E  @tab  E
-    @tab encoding and decoding supported through external library libilbc
 @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
@@ -838,22 +742,15 @@ following image formats are supported:
     @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
-    @tab libtwolame can be used alternatively for encoding.
 @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 Opus                   @tab  E  @tab  E
-    @tab supported through external library libopus
 @item PCM A-law              @tab  X  @tab  X
 @item PCM mu-law             @tab  X  @tab  X
-@item PCM signed 8-bit planar  @tab  X  @tab  X
-@item PCM signed 16-bit big-endian planar  @tab  X  @tab  X
-@item PCM signed 16-bit little-endian planar  @tab  X  @tab  X
-@item PCM signed 24-bit little-endian planar  @tab  X  @tab  X
-@item PCM signed 32-bit little-endian planar  @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
@@ -884,7 +781,6 @@ following image formats are supported:
     @tab Real 28800 bit/s codec
 @item RealAudio 3.0 (dnet)   @tab IX  @tab  X
     @tab Real low bitrate AC-3 codec
-@item RealAudio Lossless     @tab     @tab  X
 @item RealAudio SIPR / ACELP.NET @tab     @tab  X
 @item Shorten                @tab     @tab  X
 @item Sierra VMD audio       @tab     @tab  X
@@ -897,20 +793,16 @@ following image formats are supported:
     @tab experimental codec
 @item Speex                  @tab  E  @tab  E
     @tab supported through external library libspeex
-@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
 @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 VIMA                   @tab     @tab  X
-    @tab Used in LucasArts SMUSH animations.
 @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 Lossless @tab  @tab  X
 @item Windows Media Audio Pro @tab    @tab  X
 @item Windows Media Audio Voice @tab  @tab  X
 @end multitable
@@ -926,27 +818,13 @@ performance on systems without hardware floating point support).
 
 @multitable @columnfractions .4 .1 .1 .1 .1
 @item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
-@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
-@item AQTitle          @tab   @tab X @tab   @tab X
-@item DVB              @tab X @tab X @tab X @tab X
-@item DVD              @tab X @tab X @tab X @tab X
-@item JACOsub          @tab X @tab X @tab   @tab X
-@item MicroDVD         @tab X @tab X @tab   @tab X
-@item MPL2             @tab   @tab X @tab   @tab X
-@item MPsub (MPlayer)  @tab   @tab X @tab   @tab X
-@item PGS              @tab   @tab   @tab   @tab X
-@item PJS (Phoenix)    @tab   @tab X @tab   @tab X
-@item RealText         @tab   @tab X @tab   @tab X
-@item SAMI             @tab   @tab X @tab   @tab X
-@item SSA/ASS          @tab X @tab X @tab X @tab X
-@item SubRip (SRT)     @tab X @tab X @tab X @tab X
-@item SubViewer v1     @tab   @tab X @tab   @tab X
-@item SubViewer        @tab   @tab X @tab   @tab X
-@item TED Talks captions @tab @tab X @tab   @tab X
-@item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
-@item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab   @tab X @tab   @tab X
-@item XSUB             @tab   @tab   @tab X @tab X
+@item SSA/ASS      @tab X @tab X @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 MicroDVD     @tab X @tab X @tab   @tab
+@item PGS          @tab   @tab   @tab   @tab X
+@item SubRip (SRT) @tab X @tab X @tab X @tab X
+@item XSUB         @tab   @tab   @tab X @tab X
 @end multitable
 
 @code{X} means that the feature is supported.
@@ -955,31 +833,19 @@ performance on systems without hardware floating point support).
 
 @multitable @columnfractions .4 .1
 @item Name         @tab Support
+@item Apple HTTP Live Streaming @tab X
 @item file         @tab X
 @item Gopher       @tab X
-@item HLS          @tab X
 @item HTTP         @tab X
-@item HTTPS        @tab X
-@item MMSH         @tab X
-@item MMST         @tab X
+@item MMS          @tab X
 @item pipe         @tab X
-@item RTMP         @tab X
-@item RTMPE        @tab X
-@item RTMPS        @tab X
-@item RTMPT        @tab X
-@item RTMPTE       @tab X
-@item RTMPTS       @tab X
 @item RTP          @tab X
-@item SCTP         @tab X
 @item TCP          @tab X
-@item TLS          @tab X
 @item UDP          @tab X
 @end multitable
 
 @code{X} means that the protocol is supported.
 
-@code{E} means that support is provided through an external library.
-
 
 @section Input/Output Devices
 
@@ -987,17 +853,12 @@ performance on systems without hardware floating point support).
 @item Name              @tab Input  @tab Output
 @item ALSA              @tab X      @tab X
 @item BKTR              @tab X      @tab
-@item caca              @tab        @tab X
 @item DV1394            @tab X      @tab
-@item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab
 @item JACK              @tab X      @tab
-@item LIBCDIO           @tab X
 @item LIBDC1394         @tab X      @tab
-@item OpenAL            @tab X
 @item OSS               @tab X      @tab X
 @item Pulseaudio        @tab X      @tab
-@item SDL               @tab        @tab X
+@item Video4Linux       @tab X      @tab
 @item Video4Linux2      @tab X      @tab
 @item VfW capture       @tab X      @tab
 @item X11 grabbing      @tab X      @tab
@@ -1009,12 +870,11 @@ performance on systems without hardware floating point support).
 
 @multitable @columnfractions .4 .1 .1
 @item Codec/format      @tab Read   @tab Write
-@item AVI               @tab X      @tab X
 @item DV                @tab X      @tab X
 @item GXF               @tab X      @tab X
-@item MOV               @tab X      @tab X
+@item MOV               @tab X      @tab
 @item MPEG1/2           @tab X      @tab X
-@item MXF               @tab X      @tab X
+@item MXF               @tab        @tab X
 @end multitable
 
 @bye

doc/git-howto.texi

@@ -65,14 +65,6 @@ git clone git@@source.ffmpeg.org:ffmpeg <target>
 This will put the FFmpeg sources into the directory @var{<target>} and let
 you push back your changes to the remote repository.
 
-Make sure that you do not have Windows line endings in your checkouts,
-otherwise you may experience spurious compilation failures. One way to
-achieve this is to run
-
-@example
-git config --global core.autocrlf false
-@end example
-
 
 @section Updating the source tree to the latest revision
 
@@ -258,32 +250,6 @@ git commit
 @end example
 
 
-@chapter Git configuration
-
-In order to simplify a few workflows, it is advisable to configure both
-your personal Git installation and your local FFmpeg repository.
-
-@section Personal Git installation
-
-Add the following to your @file{~/.gitconfig} to help @command{git send-email}
-and @command{git format-patch} detect renames:
-
-@example
-[diff]
-        renames = copy
-@end example
-
-@section Repository configuration
-
-In order to have @command{git send-email} automatically send patches
-to the ffmpeg-devel mailing list, add the following stanza
-to @file{/path/to/ffmpeg/repository/.git/config}:
-
-@example
-[sendemail]
-        to = ffmpeg-devel@@ffmpeg.org
-@end example
-
 @chapter FFmpeg specific
 
 @section Reverting broken commits
@@ -372,43 +338,6 @@ git checkout -b svn_23456 $SHA1
 
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
-
-@chapter pre-push checklist
-
-Once you have a set of commits that you feel are ready for pushing,
-work through the following checklist to doublecheck everything is in
-proper order. This list tries to be exhaustive. In case you are just
-pushing a typo in a comment, some of the steps may be unnecessary.
-Apply your common sense, but if in doubt, err on the side of caution.
-
-First, make sure that the commits and branches you are going to push
-match what you want pushed and that nothing is missing, extraneous or
-wrong. You can see what will be pushed by running the git push command
-with --dry-run first. And then inspecting the commits listed with
-@command{git log -p 1234567..987654}. The @command{git status} command
-may help in finding local changes that have been forgotten to be added.
-
-Next let the code pass through a full run of our testsuite.
-
-@itemize
-@item @command{make distclean}
-@item @command{/path/to/ffmpeg/configure}
-@item @command{make check}
-@item if fate fails due to missing samples run @command{make fate-rsync} and retry
-@end itemize
-
-Make sure all your changes have been checked before pushing them, the
-testsuite only checks against regressions and that only to some extend. It does
-obviously not check newly added features/code to be working unless you have
-added a test for that (which is recommended).
-
-Also note that every single commit should pass the test suite, not just
-the result of a series of patches.
-
-Once everything passed, push the changes to your public ffmpeg clone and post a
-merge request to ffmpeg-devel. You can also push them directly but this is not
-recommended.
-
 @chapter Server Issues
 
 Contact the project admins @email{root@@ffmpeg.org} if you have technical

doc/indevs.texi

@@ -59,7 +59,7 @@ BSD video input device.
 
 Windows DirectShow input device.
 
-DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.
+DirectShow support is enabled when FFmpeg is built with mingw-w64.
 Currently only audio and video devices are supported.
 
 Multiple devices may be opened as separate inputs, but they may also be
@@ -112,19 +112,6 @@ defaults to 0).
 Set audio device number for devices with same name (starts at 0,
 defaults to 0).
 
-@item pixel_format
-Select pixel format to be used by DirectShow. This may only be set when
-the video codec is not set or set to rawvideo.
-
-@item audio_buffer_size
-Set audio device buffer size in milliseconds (which can directly
-impact latency, depending on the device).
-Defaults to using the audio device's
-default buffer size (typically some multiple of 500ms).
-Setting this value too low can degrade performance.
-See also
-@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx}
-
 @end table
 
 @subsection Examples
@@ -192,66 +179,6 @@ ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
 
 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
-@section iec61883
-
-FireWire DV/HDV input device using libiec61883.
-
-To enable this input device, you need libiec61883, libraw1394 and
-libavc1394 installed on your system. Use the configure option
-@code{--enable-libiec61883} to compile with the device enabled.
-
-The iec61883 capture device supports capturing from a video device
-connected via IEEE1394 (FireWire), using libiec61883 and the new Linux
-FireWire stack (juju). This is the default DV/HDV input method in Linux
-Kernel 2.6.37 and later, since the old FireWire stack was removed.
-
-Specify the FireWire port to be used as input file, or "auto"
-to choose the first port connected.
-
-@subsection Options
-
-@table @option
-
-@item dvtype
-Override autodetection of DV/HDV. This should only be used if auto
-detection does not work, or if usage of a different device type
-should be prohibited. Treating a DV device as HDV (or vice versa) will
-not work and result in undefined behavior.
-The values @option{auto}, @option{dv} and @option{hdv} are supported.
-
-@item dvbuffer
-Set maxiumum size of buffer for incoming data, in frames. For DV, this
-is an exact value. For HDV, it is not frame exact, since HDV does
-not have a fixed frame size.
-
-@item dvguid
-Select the capture device by specifying it's GUID. Capturing will only
-be performed from the specified device and fails if no device with the
-given GUID is found. This is useful to select the input if multiple
-devices are connected at the same time.
-Look at /sys/bus/firewire/devices to find out the GUIDs.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Grab and show the input of a FireWire DV/HDV device.
-@example
-ffplay -f iec61883 -i auto
-@end example
-
-@item
-Grab and record the input of a FireWire DV/HDV device,
-using a packet buffer of 100000 packets if the source is HDV.
-@example
-ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
-@end example
-
-@end itemize
-
 @section jack
 
 JACK input device.
@@ -327,12 +254,6 @@ label, but all the others need to be specified explicitly.
 
 If not specified defaults to the filename specified for the input
 device.
-
-@item graph_file
-Set the filename of the filtergraph to be read and sent to the other
-filters. Syntax of the filtergraph is the same as the one specified by
-the option @var{graph}.
-
 @end table
 
 @subsection Examples
@@ -341,14 +262,14 @@ the option @var{graph}.
 @item
 Create a color video stream and play it back with @command{ffplay}:
 @example
-ffplay -f lavfi -graph "color=c=pink [out0]" dummy
+ffplay -f lavfi -graph "color=pink [out0]" dummy
 @end example
 
 @item
 As the previous example, but use filename for specifying the graph
 description, and omit the "out0" label:
 @example
-ffplay -f lavfi color=c=pink
+ffplay -f lavfi color=pink
 @end example
 
 @item
@@ -583,9 +504,9 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
 @end example
 
-@section video4linux2
+@section video4linux and video4linux2
 
-Video4Linux2 input video device.
+Video4Linux and Video4Linux2 input video devices.
 
 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
@@ -593,28 +514,36 @@ systems tend to automatically create such nodes when the device
 kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
 the device.
 
-Video4Linux2 devices usually support a limited set of
+Video4Linux and Video4Linux2 devices only support a limited set of
 @var{width}x@var{height} sizes and framerates. You can check which are
-supported using @command{-list_formats all} for Video4Linux2 devices.
+supported for example with the command @command{dov4l} for Video4Linux
+devices and using @command{-list_formats all} for Video4Linux2 devices.
 
-Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
+If the size for the device is set to 0x0, the input device will
+try to auto-detect the size to use.
+Only for the video4linux2 device, if the frame rate is set to 0/0 the
+input device will use the frame rate value already set in the driver.
 
-The time base for the timestamps is 1 microsecond. Depending on the kernel
-version and configuration, the timestamps may be derived from the real time
-clock (origin at the Unix Epoch) or the monotonic clock (origin usually at
-boot time, unaffected by NTP or manual changes to the clock). The
-@option{-timestamps abs} or @option{-ts abs} option can be used to force
-conversion into the real time clock.
+Video4Linux support is deprecated since Linux 2.6.30, and will be
+dropped in later versions.
 
 Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
 option), it will always be used.
-@example
-# Grab and show the input of a video4linux2 device.
-ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
 
-# Grab and record the input of a video4linux2 device, leave the
-framerate and size as previously set.
-ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
+Follow some usage examples of the video4linux devices with the ff*
+tools.
+@example
+# Grab and show the input of a video4linux device, frame rate is set
+# to the default of 25/1.
+ffplay -s 320x240 -f video4linux /dev/video0
+
+# Grab and show the input of a video4linux2 device, auto-adjust size.
+ffplay -f video4linux2 /dev/video0
+
+# Grab and record the input of a video4linux2 device, auto-adjust size,
+# frame rate value defaults to 0/0 so it is read from the video4linux2
+# driver.
+ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
 @end example
 
 "v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
@@ -656,23 +585,17 @@ properties of your X11 display (e.g. grep for "name" or "dimensions").
 For example to grab from @file{:0.0} using @command{ffmpeg}:
 @example
 ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
-@end example
 
-Grab at position @code{10,20}:
-@example
+# Grab at position 10,20.
 ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
 @end example
 
-@subsection Options
+@subsection @var{follow_mouse} AVOption
 
-@table @option
-@item draw_mouse
-Specify whether to draw the mouse pointer. A value of @code{0} specify
-not to draw the pointer. Default value is @code{1}.
-
-@item follow_mouse
-Make the grabbed area follow the mouse. The argument can be
-@code{centered} or a number of pixels @var{PIXELS}.
+The syntax is:
+@example
+-follow_mouse centered|@var{PIXELS}
+@end example
 
 When it is specified with "centered", the grabbing region follows the mouse
 pointer and keeps the pointer at the center of region; otherwise, the region
@@ -682,36 +605,28 @@ zero) to the edge of region.
 For example:
 @example
 ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
-@end example
 
-To follow only when the mouse pointer reaches within 100 pixels to edge:
-@example
+# Follows only when the mouse pointer reaches within 100 pixels to edge
 ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
 @end example
 
-@item framerate
-Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a framerate of @code{30000/1001}.
+@subsection @var{show_region} AVOption
 
-@item show_region
-Show grabbed region on screen.
+The syntax is:
+@example
+-show_region 1
+@end example
 
-If @var{show_region} is specified with @code{1}, then the grabbing
-region will be indicated on screen. With this option, it is easy to
-know what is being grabbed if only a portion of the screen is grabbed.
+If @var{show_region} AVOption is specified with @var{1}, then the grabbing
+region will be indicated on screen. With this option, it's easy to know what is
+being grabbed if only a portion of the screen is grabbed.
 
 For example:
 @example
 ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
-@end example
 
-With @var{follow_mouse}:
-@example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
+# With follow_mouse
+ffmpeg -f x11grab -follow_mouse centered -show_region 1  -r 25 -s cif -i :0.0 out.mpg
 @end example
 
-@item video_size
-Set the video frame size. Default value is @code{vga}.
-@end table
-
 @c man end INPUT DEVICES

doc/issue_tracker.txt

@@ -24,7 +24,7 @@ a mail for every change to every issue.
 The subscription URL for the ffmpeg-trac list is:
 http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-http(s)://trac.ffmpeg.org
+http(s)://ffmpeg.org/trac/ffmpeg
 
 Type:
 -----

doc/libavcodec.texi

@@ -1,48 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavcodec Documentation
-@titlepage
-@center @titlefont{Libavcodec Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavcodec library provides a generic encoding/decoding framework
-and contains multiple decoders and encoders for audio, video and
-subtitle streams, and several bitstream filters.
-
-The shared architecture provides various services ranging from bit
-stream I/O to DSP optimizations, and makes it suitable for
-implementing robust and fast codecs as well as for experimentation.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
-libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavcodec
-@settitle media streams decoding and encoding library
-
-@end ignore
-
-@bye

doc/libavdevice.texi

@@ -1,45 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavdevice Documentation
-@titlepage
-@center @titlefont{Libavdevice Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavdevice library provides a generic framework for grabbing from
-and rendering to many common multimedia input/output devices, and
-supports several input and output devices, including Video4Linux2,
-VfW, DShow, and ALSA.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-devices(1),
-libavutil(3), libavcodec(3), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavdevice
-@settitle multimedia device handling library
-
-@end ignore
-
-@bye

doc/libavfilter.texi

@@ -9,36 +9,84 @@
 
 @contents
 
-@chapter Description
-@c man begin DESCRIPTION
+@chapter Introduction
 
-The libavfilter library provides a generic audio/video filtering
-framework containing several filters, sources and sinks.
+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.
 
-@c man end DESCRIPTION
+Audio filtering integration into the main FFmpeg repository is a work in
+progress, so audio API and ABI should not be considered stable yet.
 
-@chapter See Also
+@chapter Tutorial
 
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-filters.html,ffmpeg-filters},
-@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
-@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
-@end ifhtml
+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:
 
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-filters(1),
-libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
-@end ifnothtml
+@example
+input --> split --> fifo -----------------------> overlay --> output
+            |                                        ^
+            |                                        |
+            +------> fifo --> crop --> vflip --------+
+@end example
 
-@include authors.texi
+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:
 
-@ignore
+@example
+ffmpeg -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output
+@end example
 
-@setfilename libavfilter
-@settitle multimedia filtering library
+The result will be that in output the top half of the video is mirrored
+onto the bottom half.
 
-@end ignore
+Video filters are loaded using the @var{-vf} option passed to
+@command{ffmpeg} or to @command{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

doc/libavformat.texi

@@ -1,48 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavformat Documentation
-@titlepage
-@center @titlefont{Libavformat Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavformat library provides a generic framework for multiplexing
-and demultiplexing (muxing and demuxing) audio, video and subtitle
-streams. It encompasses multiple muxers and demuxers for multimedia
-container formats.
-
-It also supports several input and output protocols to access a media
-resource.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-formats(1), ffmpeg-protocols(1),
-libavutil(3), libavcodec(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavformat
-@settitle multimedia muxing and demuxing library
-
-@end ignore
-
-@bye

doc/libavutil.texi

@@ -1,44 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavutil Documentation
-@titlepage
-@center @titlefont{Libavutil Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavutil library is a utility library to aid portable
-multimedia programming. It contains safe portable string functions,
-random number generators, data structures, additional mathematics
-functions, cryptography and multimedia related functionality (like
-enumerations for pixel and sample formats).
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-utils(1)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavutil
-@settitle multimedia-biased utility library
-
-@end ignore
-
-@bye

doc/libswresample.texi

@@ -1,70 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libswresample Documentation
-@titlepage
-@center @titlefont{Libswresample Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libswresample library performs highly optimized audio resampling,
-rematrixing and sample format conversion operations.
-
-Specifically, this library performs the following conversions:
-
-@itemize
-@item
-@emph{Resampling}: is the process of changing the audio rate, for
-example from an high sample rate of 44100Hz to 8000Hz. Audio
-conversion from high to low sample rate is a lossy process. Several
-resampling options and algorithms are available.
-
-@item
-@emph{Format conversion}: is the process of converting the type of
-samples, for example from 16-bit signed samples to unsigned 8-bit or
-float samples. It also handles packing conversion, when passing from
-packed layout (all samples belonging to distinct channels interleaved
-in the same buffer), to planar layout (all samples belonging to the
-same channel stored in a dedicated buffer or "plane").
-
-@item
-@emph{Rematrixing}: is the process of changing the channel layout, for
-example from stereo to mono. When the input channels cannot be mapped
-to the output streams, the process is lossy, since it involves
-different gain factors and mixing.
-@end itemize
-
-Various other audio conversions (e.g. stretching and padding) are
-enabled through dedicated options.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-resampler(1),
-libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libswresample
-@settitle audio resampling library
-
-@end ignore
-
-@bye

doc/libswscale.texi

@@ -1,63 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libswscale Documentation
-@titlepage
-@center @titlefont{Libswscale Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libswscale library performs highly optimized image scaling and
-colorspace and pixel format conversion operations.
-
-Specifically, this library performs the following conversions:
-
-@itemize
-@item
-@emph{Rescaling}: is the process of changing the video size. Several
-rescaling options and algorithms are available. This is usually a
-lossy process.
-
-@item
-@emph{Pixel format conversion}: is the process of converting the image
-format and colorspace of the image, for example from planar YUV420P to
-RGB24 packed. It also handles packing conversion, that is converts
-from packed layout (all pixels belonging to distinct planes
-interleaved in the same buffer), to planar layout (all samples
-belonging to the same plane stored in a dedicated buffer or "plane").
-
-This is usually a lossy process in case the source and destination
-colorspaces differ.
-@end itemize
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-scaler(1),
-libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libswscale
-@settitle video scaling and pixel format conversion library
-
-@end ignore
-
-@bye

doc/mips.txt

@@ -1,67 +0,0 @@
-MIPS optimizations info
-===============================================
-
-MIPS optimizations of codecs are targeting MIPS 74k family of
-CPUs. Some of these optimizations are relying more on properties of
-this architecture and some are relying less (and can be used on most
-MIPS architectures without degradation in performance).
-
-Along with FFMPEG copyright notice, there is MIPS copyright notice in
-all the files that are created by people from MIPS Technologies.
-
-Example of copyright notice:
-===============================================
-/*
- * Copyright (c) 2012
- *      MIPS Technologies, Inc., California.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- * 1. Redistributions of source code must retain the above copyright
- *    notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- *    notice, this list of conditions and the following disclaimer in the
- *    documentation and/or other materials provided with the distribution.
- * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
- *    contributors may be used to endorse or promote products derived from
- *    this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
- * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
- * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
- * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
- * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
- * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
- * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
- * SUCH DAMAGE.
- *
- * Author:  Author Name (author_name@@mips.com)
- */
-
-Files that have MIPS copyright notice in them:
-===============================================
-* libavutil/mips/
-      float_dsp_mips.c
-      libm_mips.h
-* libavcodec/mips/
-      ac3dsp_mips.c
-      acelp_filters_mips.c
-      acelp_vectors_mips.c
-      amrwbdec_mips.c
-      amrwbdec_mips.h
-      celp_filters_mips.c
-      celp_math_mips.c
-      compute_antialias_fixed.h
-      compute_antialias_float.h
-      lsp_mips.h
-      dsputil_mips.c
-      fft_mips.c
-      fft_table.h
-      fft_init_table.c
-      fmtconvert_mips.c
-      mpegaudiodsp_mips_fixed.c
-      mpegaudiodsp_mips_float.c

doc/muxers.texi

@@ -18,23 +18,6 @@ enabled muxers.
 
 A description of some of the currently available muxers follows.
 
-@anchor{aiff}
-@section aiff
-
-Audio Interchange File Format muxer.
-
-It accepts the following options:
-
-@table @option
-@item write_id3v2
-Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
-
-@item id3v2_version
-Select ID3v2 version to write. Currently only version 3 and 4 (aka.
-ID3v2.3 and ID3v2.4) are supported. The default is version 4.
-
-@end table
-
 @anchor{crc}
 @section crc
 
@@ -73,37 +56,31 @@ See also the @ref{framecrc} muxer.
 @anchor{framecrc}
 @section framecrc
 
-Per-packet CRC (Cyclic Redundancy Check) testing format.
+Per-frame CRC (Cyclic Redundancy Check) testing format.
 
-This muxer computes and prints the Adler-32 CRC for each audio
-and video packet. By default audio frames are converted to signed
+This muxer computes and prints the Adler-32 CRC for each decoded audio
+and video frame. By default audio frames are converted to signed
 16-bit raw audio and video frames to raw video before computing the
 CRC.
 
 The output of the muxer consists of a line for each audio and video
-packet of the form:
-@example
-@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC}
-@end example
+frame of the form: @var{stream_index}, @var{frame_dts},
+@var{frame_size}, 0x@var{CRC}, where @var{CRC} is a hexadecimal
+number 0-padded to 8 digits containing the CRC of the decoded frame.
 
-@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
-CRC of the packet.
-
-For example to compute the CRC of the audio and video frames in
-@file{INPUT}, converted to raw audio and video packets, and store it
-in the file @file{out.crc}:
+For example to compute the CRC of each decoded frame in the input, and
+store it in the file @file{out.crc}:
 @example
 ffmpeg -i INPUT -f framecrc out.crc
 @end example
 
-To print the information to stdout, use the command:
+You can print the CRC of each decoded frame to stdout with the command:
 @example
 ffmpeg -i INPUT -f framecrc -
 @end example
 
-With @command{ffmpeg}, you can select the output format to which the
-audio and video frames are encoded before computing the CRC for each
-packet by specifying the audio and video codec. For example, to
+You can select the output format of each frame with @command{ffmpeg} by
+specifying the audio and video codec and format. For example, to
 compute the CRC of each decoded input audio frame converted to PCM
 unsigned 8-bit and of each decoded input video frame converted to
 MPEG-2 video, use the command:
@@ -113,98 +90,6 @@ ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
 
 See also the @ref{crc} muxer.
 
-@anchor{framemd5}
-@section framemd5
-
-Per-packet MD5 testing format.
-
-This muxer computes and prints the MD5 hash for each audio
-and video packet. By default audio frames are converted to signed
-16-bit raw audio and video frames to raw video before computing the
-hash.
-
-The output of the muxer consists of a line for each audio and video
-packet of the form:
-@example
-@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5}
-@end example
-
-@var{MD5} is a hexadecimal number representing the computed MD5 hash
-for the packet.
-
-For example to compute the MD5 of the audio and video frames in
-@file{INPUT}, converted to raw audio and video packets, and store it
-in the file @file{out.md5}:
-@example
-ffmpeg -i INPUT -f framemd5 out.md5
-@end example
-
-To print the information to stdout, use the command:
-@example
-ffmpeg -i INPUT -f framemd5 -
-@end example
-
-See also the @ref{md5} muxer.
-
-@anchor{hls}
-@section hls
-
-Apple HTTP Live Streaming muxer that segments MPEG-TS according to
-the HTTP Live Streaming specification.
-
-It creates a playlist file and numbered segment files. The output
-filename specifies the playlist filename; the segment filenames
-receive the same basename as the playlist, a sequential number and
-a .ts extension.
-
-@example
-ffmpeg -i in.nut out.m3u8
-@end example
-
-@table @option
-@item -hls_time @var{seconds}
-Set the segment length in seconds.
-@item -hls_list_size @var{size}
-Set the maximum number of playlist entries.
-@item -hls_wrap @var{wrap}
-Set the number after which index wraps.
-@item -start_number @var{number}
-Start the sequence from @var{number}.
-@end table
-
-@anchor{ico}
-@section ico
-
-ICO file muxer.
-
-Microsoft's icon file format (ICO) has some strict limitations that should be noted:
-
-@itemize
-@item
-Size cannot exceed 256 pixels in any dimension
-
-@item
-Only BMP and PNG images can be stored
-
-@item
-If a BMP image is used, it must be one of the following pixel formats:
-@example
-BMP Bit Depth      FFmpeg Pixel Format
-1bit               pal8
-4bit               pal8
-8bit               pal8
-16bit              rgb555le
-24bit              bgr24
-32bit              bgra
-@end example
-
-@item
-If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
-
-@item
-If a PNG image is used, it must use the rgba pixel format
-@end itemize
-
 @anchor{image2}
 @section image2
 
@@ -257,120 +142,24 @@ Note also that the pattern must not necessarily contain "%d" or
 ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
 @end example
 
-@table @option
-@item -start_number @var{number}
-Start the sequence from @var{number}.
-@end table
-
 The image muxer supports the .Y.U.V image file format. This format is
 special in that that each image frame consists of three files, for
 each of the YUV420P components. To read or write this image file format,
 specify the name of the '.Y' file. The muxer will automatically open the
 '.U' and '.V' files as required.
 
-@anchor{md5}
-@section md5
+@section mov
 
-MD5 testing format.
+MOV / MP4 muxer
 
-This muxer computes and prints the MD5 hash of all the input audio
-and video frames. By default audio frames are converted to signed
-16-bit raw audio and video frames to raw video before computing the
-hash.
-
-The output of the muxer consists of a single line of the form:
-MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing
-the computed MD5 hash.
-
-For example to compute the MD5 hash of the input converted to raw
-audio and video, and store it in the file @file{out.md5}:
-@example
-ffmpeg -i INPUT -f md5 out.md5
-@end example
-
-You can print the MD5 to stdout with the command:
-@example
-ffmpeg -i INPUT -f md5 -
-@end example
-
-See also the @ref{framemd5} muxer.
-
-@section MOV/MP4/ISMV
-
-The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
-file has all the metadata about all packets stored in one location
-(written at the end of the file, it can be moved to the start for
-better playback by adding @var{faststart} to the @var{movflags}, or
-using the @command{qt-faststart} tool). A fragmented
-file consists of a number of fragments, where packets and metadata
-about these packets are stored together. Writing a fragmented
-file has the advantage that the file is decodable even if the
-writing is interrupted (while a normal MOV/MP4 is undecodable if
-it is not properly finished), and it requires less memory when writing
-very long files (since writing normal MOV/MP4 files stores info about
-every single packet in memory until the file is closed). The downside
-is that it is less compatible with other applications.
-
-Fragmentation is enabled by setting one of the AVOptions that define
-how to cut the file into fragments:
+The muxer options are:
 
 @table @option
 @item -moov_size @var{bytes}
 Reserves space for the moov atom at the beginning of the file instead of placing the
 moov atom at the end. If the space reserved is insufficient, muxing will fail.
-@item -movflags frag_keyframe
-Start a new fragment at each video keyframe.
-@item -frag_duration @var{duration}
-Create fragments that are @var{duration} microseconds long.
-@item -frag_size @var{size}
-Create fragments that contain up to @var{size} bytes of payload data.
-@item -movflags frag_custom
-Allow the caller to manually choose when to cut fragments, by
-calling @code{av_write_frame(ctx, NULL)} to write a fragment with
-the packets written so far. (This is only useful with other
-applications integrating libavformat, not from @command{ffmpeg}.)
-@item -min_frag_duration @var{duration}
-Don't create fragments that are shorter than @var{duration} microseconds long.
 @end table
 
-If more than one condition is specified, fragments are cut when
-one of the specified conditions is fulfilled. The exception to this is
-@code{-min_frag_duration}, which has to be fulfilled for any of the other
-conditions to apply.
-
-Additionally, the way the output file is written can be adjusted
-through a few other options:
-
-@table @option
-@item -movflags empty_moov
-Write an initial moov atom directly at the start of the file, without
-describing any samples in it. Generally, an mdat/moov pair is written
-at the start of the file, as a normal MOV/MP4 file, containing only
-a short portion of the file. With this option set, there is no initial
-mdat atom, and the moov atom only describes the tracks but has
-a zero duration.
-
-Files written with this option set do not work in QuickTime.
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -movflags separate_moof
-Write a separate moof (movie fragment) atom for each track. Normally,
-packets for all tracks are written in a moof atom (which is slightly
-more efficient), but with this option set, the muxer writes one moof/mdat
-pair for each track, making it easier to separate tracks.
-
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -movflags faststart
-Run a second pass moving the moov atom on top of the file. This
-operation can take a while, and will not work in various situations such
-as fragmented output, thus it is not enabled by default.
-@end table
-
-Smooth Streaming content can be pushed in real time to a publishing
-point on IIS with this muxer. Example:
-@example
-ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
-@end example
-
 @section mpegts
 
 MPEG transport stream muxer.
@@ -497,7 +286,7 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
 @end example
 
-@section segment, stream_segment, ssegment
+@section segment
 
 Basic stream segmenter.
 
@@ -505,234 +294,27 @@ The segmenter muxer outputs streams to a number of separate files of nearly
 fixed duration. Output filename pattern can be set in a fashion similar to
 @ref{image2}.
 
-@code{stream_segment} is a variant of the muxer used to write to
-streaming output formats, i.e. which do not require global headers,
-and is recommended for outputting e.g. to MPEG transport stream segments.
-@code{ssegment} is a shorter alias for @code{stream_segment}.
-
-Every segment starts with a keyframe of the selected reference stream,
-which is set through the @option{reference_stream} option.
-
-Note that if you want accurate splitting for a video file, you need to
-make the input key frames correspond to the exact splitting times
-expected by the segmenter, or the segment muxer will start the new
-segment with the key frame found next after the specified start
-time.
-
+Every segment starts with a video keyframe, if a video stream is present.
 The segment muxer works best with a single constant frame rate video.
 
-Optionally it can generate a list of the created segments, by setting
-the option @var{segment_list}. The list type is specified by the
-@var{segment_list_type} option.
-
-The segment muxer supports the following options:
+Optionally it can generate a flat list of the created segments, one segment
+per line.
 
 @table @option
-@item reference_stream @var{specifier}
-Set the reference stream, as specified by the string @var{specifier}.
-If @var{specifier} is set to @code{auto}, the reference is choosen
-automatically. Otherwise it must be a stream specifier (see the ``Stream
-specifiers'' chapter in the ffmpeg manual) which specifies the
-reference stream. The default value is ``auto''.
-
 @item segment_format @var{format}
 Override the inner container format, by default it is guessed by the filename
 extension.
+@item segment_time @var{t}
+Set segment duration to @var{t} seconds.
 @item segment_list @var{name}
-Generate also a listfile named @var{name}. If not specified no
-listfile is generated.
-@item segment_list_flags @var{flags}
-Set flags affecting the segment list generation.
-
-It currently supports the following flags:
-@table @var
-@item cache
-Allow caching (only affects M3U8 list files).
-
-@item live
-Allow live-friendly file generation.
-
-This currently only affects M3U8 lists. In particular, write a fake
-EXT-X-TARGETDURATION duration field at the top of the file, based on
-the specified @var{segment_time}.
-@end table
-
-Default value is @code{cache}.
-
+Generate also a listfile named @var{name}.
 @item segment_list_size @var{size}
-Overwrite the listfile once it reaches @var{size} entries. If 0
-the listfile is never overwritten. Default value is 0.
-@item segment_list type @var{type}
-Specify the format for the segment list file.
-
-The following values are recognized:
-@table @option
-@item flat
-Generate a flat list for the created segments, one segment per line.
-
-@item csv, ext
-Generate a list for the created segments, one segment per line,
-each line matching the format (comma-separated values):
-@example
-@var{segment_filename},@var{segment_start_time},@var{segment_end_time}
-@end example
-
-@var{segment_filename} is the name of the output file generated by the
-muxer according to the provided pattern. CSV escaping (according to
-RFC4180) is applied if required.
-
-@var{segment_start_time} and @var{segment_end_time} specify
-the segment start and end time expressed in seconds.
-
-A list file with the suffix @code{".csv"} or @code{".ext"} will
-auto-select this format.
-
-@code{ext} is deprecated in favor or @code{csv}.
-
-@item m3u8
-Generate an extended M3U8 file, version 4, compliant with
-@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming-08.txt}.
-
-A list file with the suffix @code{".m3u8"} will auto-select this format.
+Overwrite the listfile once it reaches @var{size} entries.
 @end table
 
-If not specified the type is guessed from the list file name suffix.
-@item segment_time @var{time}
-Set segment duration to @var{time}. Default value is "2".
-@item segment_time_delta @var{delta}
-Specify the accuracy time when selecting the start time for a
-segment. Default value is "0".
-
-When delta is specified a key-frame will start a new segment if its
-PTS satisfies the relation:
 @example
-PTS >= start_time - time_delta
+ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
 @end example
 
-This option is useful when splitting video content, which is always
-split at GOP boundaries, in case a key frame is found just before the
-specified split time.
-
-In particular may be used in combination with the @file{ffmpeg} option
-@var{force_key_frames}. The key frame times specified by
-@var{force_key_frames} may not be set accurately because of rounding
-issues, with the consequence that a key frame time may result set just
-before the specified time. For constant frame rate videos a value of
-1/2*@var{frame_rate} should address the worst case mismatch between
-the specified time and the time set by @var{force_key_frames}.
-
-@item segment_times @var{times}
-Specify a list of split points. @var{times} contains a list of comma
-separated duration specifications, in increasing order.
-
-@item segment_frames @var{frames}
-Specify a list of split video frame numbers. @var{frames} contains a
-list of comma separated integer numbers, in increasing order.
-
-This option specifies to start a new segment whenever a reference
-stream key frame is found and the sequential number (starting from 0)
-of the frame is greater or equal to the next value in the list.
-
-@item segment_wrap @var{limit}
-Wrap around segment index once it reaches @var{limit}.
-
-@item segment_start_number @var{number}
-Set the sequence number of the first segment. Defaults to @code{0}.
-
-@item reset_timestamps @var{1|0}
-Reset timestamps at the begin of each segment, so that each segment
-will start with near-zero timestamps. It is meant to ease the playback
-of the generated segments. May not work with some combinations of
-muxers/codecs. It is set to @code{0} by default.
-@end table
-
-@section Examples
-
-@itemize
-@item
-To remux the content of file @file{in.mkv} to a list of segments
-@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
-generated segments to @file{out.list}:
-@example
-ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut
-@end example
-
-@item
-As the example above, but segment the input file according to the split
-points specified by the @var{segment_times} option:
-@example
-ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
-@end example
-
-@item
-As the example above, but use the @code{ffmpeg} @var{force_key_frames}
-option to force key frames in the input at the specified location, together
-with the segment option @var{segment_time_delta} to account for
-possible roundings operated when setting key frame times.
-@example
-ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
--f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut
-@end example
-In order to force key frames on the input file, transcoding is
-required.
-
-@item
-Segment the input file by splitting the input file according to the
-frame numbers sequence specified with the @var{segment_frames} option:
-@example
-ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
-@end example
-
-@item
-To convert the @file{in.mkv} to TS segments using the @code{libx264}
-and @code{libfaac} encoders:
-@example
-ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
-@end example
-
-@item
-Segment the input file, and create an M3U8 live playlist (can be used
-as live HLS source):
-@example
-ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
--segment_list_flags +live -segment_time 10 out%03d.mkv
-@end example
-@end itemize
-
-@section mp3
-
-The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
-optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
-not written by default, but may be enabled with the @code{write_id3v1} option.
-
-For seekable output the muxer also writes a Xing frame at the beginning, which
-contains the number of frames in the file. It is useful for computing duration
-of VBR files.
-
-The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
-are supplied to the muxer in form of a video stream with a single packet. There
-can be any number of those streams, each will correspond to a single APIC frame.
-The stream metadata tags @var{title} and @var{comment} map to APIC
-@var{description} and @var{picture type} respectively. See
-@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
-
-Note that the APIC frames must be written at the beginning, so the muxer will
-buffer the audio frames until it gets all the pictures. It is therefore advised
-to provide the pictures as soon as possible to avoid excessive buffering.
-
-Examples:
-
-Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
-@example
-ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
-@end example
-
-To attach a picture to an mp3 file select both the audio and the picture stream
-with @code{map}:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
--metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
-@end example
 
 @c man end MUXERS

doc/nut.texi

@@ -1,138 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle NUT
-
-@titlepage
-@center @titlefont{NUT}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-NUT is a low overhead generic container format. It stores audio, video,
-subtitle and user-defined streams in a simple, yet efficient, way.
-
-It was created by a group of FFmpeg and MPlayer developers in 2003
-and was finalized in 2008.
-
-The official nut specification is at svn://svn.mplayerhq.hu/nut
-In case of any differences between this text and the official specification,
-the official specification shall prevail.
-
-@chapter Container-specific codec tags
-
-@section Generic raw YUVA formats
-
-Since many exotic planar YUVA pixel formats are not considered by
-the AVI/QuickTime FourCC lists, the following scheme is adopted for
-representing them.
-
-The first two bytes can contain the values:
-Y1 = only Y
-Y2 = Y+A
-Y3 = YUV
-Y4 = YUVA
-
-The third byte represents the width and height chroma subsampling
-values for the UV planes, that is the amount to shift the luma
-width/height right to find the chroma width/height.
-
-The fourth byte is the number of bits used (8, 16, ...).
-
-If the order of bytes is inverted, that means that each component has
-to be read big-endian.
-
-@section Raw Audio
-
-@multitable @columnfractions .4 .4
-@item ALAW  @tab A-LAW
-@item ULAW  @tab MU-LAW
-@item P<type><interleaving><bits> @tab little-endian PCM
-@item <bits><interleaving><type>P @tab big-endian PCM
-@end multitable
-
-<type> is S for signed integer, U for unsigned integer, F for IEEE float
-<interleaving> is D for default, P is for planar.
-<bits> is 8/16/24/32
-
-@example
-PFD[32]   would for example be signed 32 bit little-endian IEEE float
-@end example
-
-@section Subtitles
-
-@multitable @columnfractions .4 .4
-@item UTF8   @tab Raw UTF-8
-@item SSA[0] @tab SubStation Alpha
-@item DVDS   @tab DVD subtitles
-@item DVBS   @tab DVB subtitles
-@end multitable
-
-@section Raw Data
-
-@multitable @columnfractions .4 .4
-@item UTF8   @tab Raw UTF-8
-@end multitable
-
-@section Codecs
-
-@multitable @columnfractions .4 .4
-@item 3IV1 @tab non-compliant MPEG-4 generated by old 3ivx
-@item ASV1 @tab Asus Video
-@item ASV2 @tab Asus Video 2
-@item CVID @tab Cinepak
-@item CYUV @tab Creative YUV
-@item DIVX @tab non-compliant MPEG-4 generated by old DivX
-@item DUCK @tab Truemotion 1
-@item FFV1 @tab FFmpeg video 1
-@item FFVH @tab FFmpeg Huffyuv
-@item H261 @tab ITU H.261
-@item H262 @tab ITU H.262
-@item H263 @tab ITU H.263
-@item H264 @tab ITU H.264
-@item HFYU @tab Huffyuv
-@item I263 @tab Intel H.263
-@item IV31 @tab Indeo 3.1
-@item IV32 @tab Indeo 3.2
-@item IV50 @tab Indeo 5.0
-@item LJPG @tab ITU JPEG (lossless)
-@item MJLS @tab ITU JPEG-LS
-@item MJPG @tab ITU JPEG
-@item MPG4 @tab MS MPEG-4v1 (not ISO MPEG-4)
-@item MP42 @tab MS MPEG-4v2
-@item MP43 @tab MS MPEG-4v3
-@item MP4V @tab ISO MPEG-4 Part 2 Video (from old encoders)
-@item mpg1 @tab ISO MPEG-1 Video
-@item mpg2 @tab ISO MPEG-2 Video
-@item MRLE @tab MS RLE
-@item MSVC @tab MS Video 1
-@item RT21 @tab Indeo 2.1
-@item RV10 @tab RealVideo 1.0
-@item RV20 @tab RealVideo 2.0
-@item RV30 @tab RealVideo 3.0
-@item RV40 @tab RealVideo 4.0
-@item SNOW @tab FFmpeg Snow
-@item SVQ1 @tab Sorenson Video 1
-@item SVQ3 @tab Sorenson Video 3
-@item theo @tab Xiph Theora
-@item TM20 @tab Truemotion 2.0
-@item UMP4 @tab non-compliant MPEG-4 generated by UB Video MPEG-4
-@item VCR1 @tab ATI VCR1
-@item VP30 @tab VP 3.0
-@item VP31 @tab VP 3.1
-@item VP50 @tab VP 5.0
-@item VP60 @tab VP 6.0
-@item VP61 @tab VP 6.1
-@item VP62 @tab VP 6.2
-@item VP70 @tab VP 7.0
-@item WMV1 @tab MS WMV7
-@item WMV2 @tab MS WMV8
-@item WMV3 @tab MS WMV9
-@item WV1F @tab non-compliant MPEG-4 generated by ?
-@item WVC1 @tab VC-1
-@item XVID @tab non-compliant MPEG-4 generated by old Xvid
-@item XVIX @tab non-compliant MPEG-4 generated by old Xvid with interlacing bug
-@end multitable
-

doc/optimization.txt

@@ -253,7 +253,7 @@ 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 MMX 2 Coprocessor: Programmers Reference Manual
+Intel Wireless MMX2 Coprocessor: Programmers Reference Manual
 http://download.intel.com/design/intelxscale/31451001.pdf
 
 PowerPC-specific:

doc/outdevs.texi

@@ -22,88 +22,6 @@ A description of the currently available output devices follows.
 
 ALSA (Advanced Linux Sound Architecture) output device.
 
-@section caca
-
-CACA output device.
-
-This output devices allows to show a video stream in CACA window.
-Only one CACA window is allowed per application, so you can
-have only one instance of this output device in an application.
-
-To enable this output device you need to configure FFmpeg with
-@code{--enable-libcaca}.
-libcaca is a graphics library that outputs text instead of pixels.
-
-For more information about libcaca, check:
-@url{http://caca.zoy.org/wiki/libcaca}
-
-@subsection Options
-
-@table @option
-
-@item window_title
-Set the CACA window title, if not specified default to the filename
-specified for the output device.
-
-@item window_size
-Set the CACA window size, can be a string of the form
-@var{width}x@var{height} or a video size abbreviation.
-If not specified it defaults to the size of the input video.
-
-@item driver
-Set display driver.
-
-@item algorithm
-Set dithering algorithm. Dithering is necessary
-because the picture being rendered has usually far more colours than
-the available palette.
-The accepted values are listed with @code{-list_dither algorithms}.
-
-@item antialias
-Set antialias method. Antialiasing smoothens the rendered
-image and avoids the commonly seen staircase effect.
-The accepted values are listed with @code{-list_dither antialiases}.
-
-@item charset
-Set which characters are going to be used when rendering text.
-The accepted values are listed with @code{-list_dither charsets}.
-
-@item color
-Set color to be used when rendering text.
-The accepted values are listed with @code{-list_dither colors}.
-
-@item list_drivers
-If set to @option{true}, print a list of available drivers and exit.
-
-@item list_dither
-List available dither options related to the argument.
-The argument must be one of @code{algorithms}, @code{antialiases},
-@code{charsets}, @code{colors}.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-The following command shows the @command{ffmpeg} output is an
-CACA window, forcing its size to 80x25:
-@example
-ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
-@end example
-
-@item
-Show the list of available drivers and exit:
-@example
-ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
-@end example
-
-@item
-Show the list of available dither colors and exit:
-@example
-ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
-@end example
-@end itemize
-
 @section oss
 
 OSS (Open Sound System) output device.
@@ -137,8 +55,7 @@ to the same value of @var{window_title}.
 @item window_size
 Set the SDL window size, can be a string of the form
 @var{width}x@var{height} or a video size abbreviation.
-If not specified it defaults to the size of the input video,
-downscaled according to the aspect ratio.
+If not specified it defaults to the size of the input video.
 @end table
 
 @subsection Examples

doc/platform.texi

@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle Platform Specific Information
+@settitle Platform Specific information
 @titlepage
-@center @titlefont{Platform Specific Information}
+@center @titlefont{Platform Specific information}
 @end titlepage
 
 @top
@@ -27,11 +27,11 @@ to configure.
 @section BSD
 
 BSD make will not build FFmpeg, you need to install and use GNU Make
-(@command{gmake}).
+(@file{gmake}).
 
 @section (Open)Solaris
 
-GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}),
+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
@@ -51,9 +51,8 @@ The toolchain provided with Xcode is sufficient to build the basic
 unacelerated code.
 
 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
-@url{https://github.com/FFmpeg/gas-preprocessor} or
 @url{http://github.com/yuvi/gas-preprocessor} to build the optimized
-assembler functions. Put the Perl script somewhere
+assembler functions. Just download the Perl script and put it somewhere
 in your PATH, FFmpeg's configure will pick it up automatically.
 
 Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
@@ -78,15 +77,30 @@ For information about compiling FFmpeg on OS/2 see
 @chapter Windows
 
 To get help and instructions for building FFmpeg under Windows, check out
-the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
+the FFmpeg Windows Help Forum at
+@url{http://ffmpeg.arrozcru.org/}.
 
-@section Native Windows compilation using MinGW or MinGW-w64
+@section Native Windows compilation
 
-FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64
-toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from
-@url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}.
-You can find detailed installation instructions in the download section and
-the FAQ.
+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}).
 
 Notes:
 
@@ -95,11 +109,14 @@ Notes:
 @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 during
+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 @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed.
+of @uref{http://www.libsdl.org/, SDL}.
+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 the FFmpeg libraries (e.g. libavutil, libavcodec,
@@ -107,105 +124,149 @@ libavformat) as DLLs.
 
 @end itemize
 
-@section Microsoft Visual C++
+@section Microsoft Visual C++ compatibility
 
-FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and
-wrapper.
+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.
 
-You will need the following prerequisites:
+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.
 
-@itemize
-@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
-@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
-@item @uref{http://www.mingw.org/, MSYS}
-@item @uref{http://yasm.tortall.net/, YASM}
-@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
-you want to run @uref{fate.html, FATE}.
-@end itemize
+@subsection Using static libraries
 
-To set up a proper MSVC environment in MSYS, you simply need to run
-@code{msys.bat} from the Visual Studio command prompt.
+Assuming you have just built and installed FFmpeg in @file{/usr/local}.
 
-Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
-somewhere in your @code{PATH}.
+@enumerate
 
-Next, make sure @code{inttypes.h} and any other headers and libs you want to use
-are located in a spot that MSVC can see. Do so by modifying the @code{LIB} and
-@code{INCLUDE} environment variables to include the @strong{Windows} paths to
-these directories. Alternatively, you can try and use the
-@code{--extra-cflags}/@code{--extra-ldflags} configure options.
+@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.
 
-Finally, run:
+@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
+
+@subsection 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
-./configure --toolchain=msvc
+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
 make
 make install
 @end example
 
-If you wish to compile shared libraries, add @code{--enable-shared} to your
-configure options. Note that due to the way MSVC handles DLL imports and
-exports, you cannot compile static and shared libraries at the same time, and
-enabling shared libraries will automatically disable the static ones.
+Your install path (@file{/usr/local/} by default) should now have the
+necessary DLL and LIB files under the @file{bin} directory.
 
-Notes:
-
-@itemize
-
-@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
-You can find out by running @code{which link} to see which @code{link.exe} you
-are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
-in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
-@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
-
-@item If you wish to build with zlib support, you will have to grab a compatible
-zlib binary from somewhere, with an MSVC import lib, or if you wish to link
-statically, you can follow the instructions below to build a compatible
-@code{zlib.lib} with MSVC. Regardless of which method you use, you must still
-follow step 3, or compilation will fail.
-@enumerate
-@item Grab the @uref{http://zlib.net/, zlib sources}.
-@item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since
-this is how FFmpeg is built as well.
-@item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets
-erroneously included when building FFmpeg.
-@item Run @code{nmake -f win32/Makefile.msc}.
-@item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC
-can see.
 @end enumerate
 
-@item FFmpeg has been tested with Visual Studio 2010 and 2012, Pro and Express.
-Anything else is not officially supported.
+Alternatively, build the libraries with a cross compiler, according to
+the instructions below in @ref{Cross compilation for Windows with Linux}.
 
-@end itemize
-
-@subsection Linking to FFmpeg with Microsoft Visual C++
-
-If you plan to link with MSVC-built static libraries, you will need
-to make sure you have @code{Runtime Library} set to
-@code{Multi-threaded (/MT)} in your project's settings.
+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 the static
+libraries (@file{libxxx.a} files) you should add the MSVC import libraries
+(@file{avcodec.lib}, @file{avformat.lib}, and
+@file{avutil.lib}). Note that you should not use the GCC import
+libraries (@file{libxxx.dll.a} files), as these will give you undefined
+reference errors. 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.
 
 FFmpeg headers do not declare global data for Windows DLLs through the usual
 dllexport/dllimport interface. Such data will be exported properly while
-building, but to use them in your MSVC code you will have to edit the
+building, but to use them in your MSVC++ code you will have to edit the
 appropriate headers and mark the data as dllimport. For example, in
 libavutil/pixdesc.h you should have:
 @example
 extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
 @end example
 
-You will also need to define @code{inline} to something MSVC understands:
-@example
-#define inline __inline
-@end example
-
-Also note, that as stated in @strong{Microsoft Visual C++}, you will need
-an MSVC-compatible @uref{http://code.google.com/p/msinttypes/, inttypes.h}.
-
-If you plan on using import libraries created by dlltool, you must
-set @code{References} to @code{No (/OPT:NOREF)} under the linker optimization
-settings, otherwise the resulting binaries will fail during runtime.
-This is not required when using import libraries generated by @code{lib.exe}.
+Note that using import libraries created by dlltool requires
+the linker optimization option to be set to
+"References: Keep Unreferenced Data (@code{/OPT:NOREF})", otherwise
+the resulting binaries will fail during runtime. This isn't
+required when using import libraries generated by lib.exe.
 This issue is reported upstream at
 @url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}.
 
@@ -214,24 +275,27 @@ To create import libraries that work with the @code{/OPT:REF} option
 
 @enumerate
 
-@item Open the @emph{Visual Studio Command Prompt}.
+@item Open @file{Visual Studio 2005 Command Prompt}.
 
 Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
 which sets up the environment variables for the Visual C++ tools
-(the standard location for this file is something like
-@file{C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat}).
+(the standard location for this file is
+@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat}).
 
 @item Enter the @file{bin} directory where the created LIB and DLL files
 are stored.
 
-@item Generate new import libraries with @command{lib.exe}:
+@item Generate new import libraries with @file{lib.exe}:
 
 @example
-lib /machine:i386 /def:..\lib\foo-version.def  /out:foo.lib
+lib /machine:i386 /def:..\lib\avcodec-53.def  /out:avcodec.lib
+lib /machine:i386 /def:..\lib\avdevice-53.def /out:avdevice.lib
+lib /machine:i386 /def:..\lib\avfilter-2.def  /out:avfilter.lib
+lib /machine:i386 /def:..\lib\avformat-53.def /out:avformat.lib
+lib /machine:i386 /def:..\lib\avutil-51.def   /out:avutil.lib
+lib /machine:i386 /def:..\lib\swscale-2.def   /out:swscale.lib
 @end example
 
-Replace @code{foo-version} and @code{foo} with the respective library names.
-
 @end enumerate
 
 @anchor{Cross compilation for Windows with Linux}
@@ -260,9 +324,24 @@ following "Devel" ones:
 binutils, gcc4-core, make, git, mingw-runtime, texi2html
 @end example
 
-In order to run FATE you will also need the following "Utils" packages:
+And the following "Utils" one:
 @example
-bc, diffutils
+diffutils
+@end example
+
+Then run
+
+@example
+./configure
+@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
@@ -275,12 +354,16 @@ These library packages are only available from
 @uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
 
 @example
-yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
-libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
+yasm, libSDL-devel, libdirac-devel, libfaac-devel, libaacplus-devel, libgsm-devel,
+libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel,
+libxvidcore-devel
 @end example
 
-The recommendation for x264 is to build it from source, as it evolves too
-quickly for Cygwin Ports to be up to date.
+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.
 
 @section Crosscompilation for Windows under Cygwin
 
@@ -304,67 +387,4 @@ and for a build with shared libraries
 ./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
 @end example
 
-@chapter Plan 9
-
-The native @uref{http://plan9.bell-labs.com/plan9/, Plan 9} compiler
-does not implement all the C99 features needed by FFmpeg so the gcc
-port must be used.  Furthermore, a few items missing from the C
-library and shell environment need to be fixed.
-
-@itemize
-
-@item GNU awk, grep, make, and sed
-
-Working packages of these tools can be found at
-@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}.
-They can be installed with @uref{http://9front.org/, 9front's} @code{pkg}
-utility by setting @code{pkgpath} to
-@code{http://ports2plan9.googlecode.com/files/}.
-
-@item Missing/broken @code{head} and @code{printf} commands
-
-Replacements adequate for building FFmpeg can be found in the
-@code{compat/plan9} directory.  Place these somewhere they will be
-found by the shell.  These are not full implementations of the
-commands and are @emph{not} suitable for general use.
-
-@item Missing C99 @code{stdint.h} and @code{inttypes.h}
-
-Replacement headers are available from
-@url{http://code.google.com/p/plan9front/issues/detail?id=152}.
-
-@item Missing or non-standard library functions
-
-Some functions in the C library are missing or incomplete.  The
-@code{@uref{http://ports2plan9.googlecode.com/files/gcc-apelibs-1207.tbz,
-gcc-apelibs-1207}} package from
-@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}
-includes an updated C library, but installing the full package gives
-unusable executables.  Instead, keep the files from @code{gccbin.tgz}
-under @code{/386/lib/gnu}.  From the @code{libc.a} archive in the
-@code{gcc-apelibs-1207} package, extract the following object files and
-turn them into a library:
-
-@itemize
-@item @code{strerror.o}
-@item @code{strtoll.o}
-@item @code{snprintf.o}
-@item @code{vsnprintf.o}
-@item @code{vfprintf.o}
-@item @code{_IO_getc.o}
-@item @code{_IO_putc.o}
-@end itemize
-
-Use the @code{--extra-libs} option of @code{configure} to inform the
-build system of this library.
-
-@item FPU exceptions enabled by default
-
-Unlike most other systems, Plan 9 enables FPU exceptions by default.
-These must be disabled before calling any FFmpeg functions.  While the
-included tools will do this automatically, other users of the
-libraries must do it themselves.
-
-@end itemize
-
 @bye

doc/print_options.c

@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2012 Anton Khirnov
- *
- * This file is part of Libav.
- *
- * Libav is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * Libav 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * generate texinfo manpages for avoptions
- */
-
-#include <stddef.h>
-#include <string.h>
-#include <float.h>
-
-#include "libavformat/avformat.h"
-#include "libavcodec/avcodec.h"
-#include "libavutil/opt.h"
-
-static void print_usage(void)
-{
-    fprintf(stderr, "Usage: enum_options type\n"
-            "type: format codec\n");
-    exit(1);
-}
-
-static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
-{
-    printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
-    switch (o->type) {
-    case AV_OPT_TYPE_BINARY:   printf("hexadecimal string"); break;
-    case AV_OPT_TYPE_STRING:   printf("string");             break;
-    case AV_OPT_TYPE_INT:
-    case AV_OPT_TYPE_INT64:    printf("integer");            break;
-    case AV_OPT_TYPE_FLOAT:
-    case AV_OPT_TYPE_DOUBLE:   printf("float");              break;
-    case AV_OPT_TYPE_RATIONAL: printf("rational number");    break;
-    case AV_OPT_TYPE_FLAGS:    printf("flags");              break;
-    default:                   printf("value");              break;
-    }
-    printf("} (@emph{");
-
-    if (o->flags & AV_OPT_FLAG_DECODING_PARAM) {
-        printf("input");
-        if (o->flags & AV_OPT_FLAG_ENCODING_PARAM)
-            printf("/");
-    }
-    if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) printf("output");
-    if (o->flags & AV_OPT_FLAG_AUDIO_PARAM)    printf(",audio");
-    if (o->flags & AV_OPT_FLAG_VIDEO_PARAM)    printf(",video");
-    if (o->flags & AV_OPT_FLAG_SUBTITLE_PARAM) printf(",subtitles");
-
-    printf("})\n");
-    if (o->help)
-        printf("%s\n", o->help);
-
-    if (o->unit) {
-        const AVOption *u;
-        printf("\nPossible values:\n@table @samp\n");
-
-        for (u = opts; u->name; u++) {
-            if (u->type == AV_OPT_TYPE_CONST && u->unit && !strcmp(u->unit, o->unit))
-                printf("@item %s\n%s\n", u->name, u->help ? u->help : "");
-        }
-        printf("@end table\n");
-    }
-}
-
-static void show_opts(const AVOption *opts, int per_stream)
-{
-    const AVOption *o;
-
-    printf("@table @option\n");
-    for (o = opts; o->name; o++) {
-        if (o->type != AV_OPT_TYPE_CONST)
-            print_option(opts, o, per_stream);
-    }
-    printf("@end table\n");
-}
-
-static void show_format_opts(void)
-{
-#include "libavformat/options_table.h"
-
-    printf("@section Format AVOptions\n");
-    show_opts(options, 0);
-}
-
-static void show_codec_opts(void)
-{
-#include "libavcodec/options_table.h"
-
-    printf("@section Codec AVOptions\n");
-    show_opts(options, 1);
-}
-
-int main(int argc, char **argv)
-{
-    if (argc < 2)
-        print_usage();
-
-    printf("@c DO NOT EDIT THIS FILE!\n"
-           "@c It was generated by print_options.\n\n");
-    if (!strcmp(argv[1], "format"))
-        show_format_opts();
-    else if (!strcmp(argv[1], "codec"))
-        show_codec_opts();
-    else
-        print_usage();
-
-    return 0;
-}

doc/protocols.texi

@@ -19,34 +19,20 @@ supported protocols.
 
 A description of the currently available protocols follows.
 
-@section bluray
+@section applehttp
 
-Read BluRay playlist.
+Read Apple HTTP Live Streaming compliant segmented stream as
+a uniform one. The M3U8 playlists describing the segments can be
+remote HTTP resources or local files, accessed using the standard
+file protocol.
+HTTP is default, specific protocol can be declared by specifying
+"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
+is either "file" or "http".
 
-The accepted options are:
-@table @option
-
-@item angle
-BluRay angle
-
-@item chapter
-Start chapter (1...N)
-
-@item playlist
-Playlist to read (BDMV/PLAYLIST/?????.mpls)
-
-@end table
-
-Examples:
-
-Read longest playlist from BluRay mounted to /mnt/bluray:
 @example
-bluray:/mnt/bluray
-@end example
-
-Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
-@example
--playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
+applehttp://host/path/to/remote/resource.m3u8
+applehttp+http://host/path/to/remote/resource.m3u8
+applehttp+file://path/to/local/resource.m3u8
 @end example
 
 @section concat
@@ -75,15 +61,6 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 Note that you may need to escape the character "|" which is special for
 many shells.
 
-@section data
-
-Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
-
-For example, to convert a GIF file given inline with @command{ffmpeg}:
-@example
-ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
-@end example
-
 @section file
 
 File access protocol.
@@ -104,26 +81,6 @@ specified with the name "FILE.mpeg" is interpreted as the URL
 
 Gopher protocol.
 
-@section hls
-
-Read Apple HTTP Live Streaming compliant segmented stream as
-a uniform one. The M3U8 playlists describing the segments can be
-remote HTTP resources or local files, accessed using the standard
-file protocol.
-The nested protocol is declared by specifying
-"+@var{proto}" after the hls URI scheme name, where @var{proto}
-is either "file" or "http".
-
-@example
-hls+http://host/path/to/remote/resource.m3u8
-hls+file://path/to/local/resource.m3u8
-@end example
-
-Using this protocol is discouraged - the hls demuxer should work
-just as well (if not, please report the issues) and is more complete.
-To use the hls demuxer instead, simply use the direct URLs to the
-m3u8 files.
-
 @section http
 
 HTTP (Hyper Text Transfer Protocol).
@@ -203,7 +160,7 @@ content across a TCP/IP network.
 
 The required syntax is:
 @example
-rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
+rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
 @end example
 
 The accepted parameters are:
@@ -218,88 +175,11 @@ The number of the TCP port to use (by default is 1935).
 @item app
 It is the name of the application to access. It usually corresponds to
 the path where the application is installed on the RTMP server
-(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
-the value parsed from the URI through the @code{rtmp_app} option, too.
+(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
 
 @item playpath
 It is the path or name of the resource to play with reference to the
-application specified in @var{app}, may be prefixed by "mp4:". You
-can override the value parsed from the URI through the @code{rtmp_playpath}
-option, too.
-
-@item listen
-Act as a server, listening for an incoming connection.
-
-@item timeout
-Maximum time to wait for the incoming connection. Implies listen.
-@end table
-
-Additionally, the following parameters can be set via command line options
-(or in code via @code{AVOption}s):
-@table @option
-
-@item rtmp_app
-Name of application to connect on the RTMP server. This option
-overrides the parameter specified in the URI.
-
-@item rtmp_buffer
-Set the client buffer time in milliseconds. The default is 3000.
-
-@item rtmp_conn
-Extra arbitrary AMF connection parameters, parsed from a string,
-e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
-Each value is prefixed by a single character denoting the type,
-B for Boolean, N for number, S for string, O for object, or Z for null,
-followed by a colon. For Booleans the data must be either 0 or 1 for
-FALSE or TRUE, respectively.  Likewise for Objects the data must be 0 or
-1 to end or begin an object, respectively. Data items in subobjects may
-be named, by prefixing the type with 'N' and specifying the name before
-the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
-times to construct arbitrary AMF sequences.
-
-@item rtmp_flashver
-Version of the Flash plugin used to run the SWF player. The default
-is LNX 9,0,124,2.
-
-@item rtmp_flush_interval
-Number of packets flushed in the same request (RTMPT only). The default
-is 10.
-
-@item rtmp_live
-Specify that the media is a live stream. No resuming or seeking in
-live streams is possible. The default value is @code{any}, which means the
-subscriber first tries to play the live stream specified in the
-playpath. If a live stream of that name is not found, it plays the
-recorded stream. The other possible values are @code{live} and
-@code{recorded}.
-
-@item rtmp_pageurl
-URL of the web page in which the media was embedded. By default no
-value will be sent.
-
-@item rtmp_playpath
-Stream identifier to play or to publish. This option overrides the
-parameter specified in the URI.
-
-@item rtmp_subscribe
-Name of live stream to subscribe to. By default no value will be sent.
-It is only sent if the option is specified or if rtmp_live
-is set to live.
-
-@item rtmp_swfhash
-SHA256 hash of the decompressed SWF file (32 bytes).
-
-@item rtmp_swfsize
-Size of the decompressed SWF file, required for SWFVerification.
-
-@item rtmp_swfurl
-URL of the SWF player for the media. By default no value will be sent.
-
-@item rtmp_swfverify
-URL to player swf file, compute hash/size automatically.
-
-@item rtmp_tcurl
-URL of the target stream. Defaults to proto://host[:port]/app.
+application specified in @var{app}, may be prefixed by "mp4:".
 
 @end table
 
@@ -309,46 +189,6 @@ For example to read with @command{ffplay} a multimedia resource named
 ffplay rtmp://myserver/vod/sample
 @end example
 
-@section rtmpe
-
-Encrypted Real-Time Messaging Protocol.
-
-The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
-streaming multimedia content within standard cryptographic primitives,
-consisting of Diffie-Hellman key exchange and HMACSHA256, generating
-a pair of RC4 keys.
-
-@section rtmps
-
-Real-Time Messaging Protocol over a secure SSL connection.
-
-The Real-Time Messaging Protocol (RTMPS) is used for streaming
-multimedia content across an encrypted connection.
-
-@section rtmpt
-
-Real-Time Messaging Protocol tunneled through HTTP.
-
-The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
-for streaming multimedia content within HTTP requests to traverse
-firewalls.
-
-@section rtmpte
-
-Encrypted Real-Time Messaging Protocol tunneled through HTTP.
-
-The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
-is used for streaming multimedia content within HTTP requests to traverse
-firewalls.
-
-@section rtmpts
-
-Real-Time Messaging Protocol tunneled through HTTPS.
-
-The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
-for streaming multimedia content within HTTPS requests to traverse
-firewalls.
-
 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
 
 Real-Time Messaging Protocol and its variants supported through
@@ -441,14 +281,12 @@ Flags for @code{rtsp_flags}:
 @table @option
 @item filter_src
 Accept packets only from negotiated peer address and port.
-@item listen
-Act as a server, listening for an incoming connection.
 @end table
 
 When receiving data over UDP, the demuxer tries to reorder received packets
-(since they may arrive out of order, or packets may get lost totally). This
-can be disabled by setting the maximum demuxing delay to zero (via
-the @code{max_delay} field of AVFormatContext).
+(since they may arrive out of order, or packets may get lost totally). In
+order for this to be enabled, a maximum delay must be specified in the
+@code{max_delay} field of AVFormatContext.
 
 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
 streams to display can be chosen with @code{-vst} @var{n} and
@@ -475,12 +313,6 @@ To send a stream in realtime to a RTSP server, for others to watch:
 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
 @end example
 
-To receive a stream in realtime:
-
-@example
-ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
-@end example
-
 @section sap
 
 Session Announcement Protocol (RFC 2974). This is not technically a
@@ -587,11 +419,6 @@ tcp://@var{hostname}:@var{port}[?@var{options}]
 @item listen
 Listen for an incoming connection
 
-@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
-In write mode: if socket cannot be written in more than this time interval, raise error.
-This also sets timeout on TCP connection establishing.
-
 @example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
@@ -599,48 +426,6 @@ ffplay tcp://@var{hostname}:@var{port}
 
 @end table
 
-@section tls
-
-Transport Layer Security/Secure Sockets Layer
-
-The required syntax for a TLS/SSL url is:
-@example
-tls://@var{hostname}:@var{port}[?@var{options}]
-@end example
-
-@table @option
-
-@item listen
-Act as a server, listening for an incoming connection.
-
-@item cafile=@var{filename}
-Certificate authority file. The file must be in OpenSSL PEM format.
-
-@item cert=@var{filename}
-Certificate file. The file must be in OpenSSL PEM format.
-
-@item key=@var{filename}
-Private key file.
-
-@item verify=@var{0|1}
-Verify the peer's certificate.
-
-@end table
-
-Example command lines:
-
-To create a TLS/SSL server that serves an input stream.
-
-@example
-ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
-@end example
-
-To play back a stream from the TLS/SSL server using @command{ffplay}:
-
-@example
-ffplay tls://@var{hostname}:@var{port}
-@end example
-
 @section udp
 
 User Datagram Protocol.
@@ -650,23 +435,16 @@ The required syntax for a UDP url is:
 udp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
-@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
-
-In case threading is enabled on the system, a circular buffer is used
-to store the incoming data, which allows to reduce loss of data due to
-UDP socket buffer overruns. The @var{fifo_size} and
-@var{overrun_nonfatal} options are related to this buffer.
-
-The list of supported options follows.
+@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
+Follow the list of supported options.
 
 @table @option
 
 @item buffer_size=@var{size}
-Set the UDP socket buffer size in bytes. This is used both for the
-receiving and the sending buffer size.
+set the UDP buffer size in bytes
 
 @item localport=@var{port}
-Override the local UDP port to bind with.
+override the local UDP port to bind with
 
 @item localaddr=@var{addr}
 Choose the local IP address. This is useful e.g. if sending multicast
@@ -674,13 +452,13 @@ and the host has multiple interfaces, where the user can choose
 which interface to send on by specifying the IP address of that interface.
 
 @item pkt_size=@var{size}
-Set the size in bytes of UDP packets.
+set the size in bytes of UDP packets
 
 @item reuse=@var{1|0}
-Explicitly allow or disallow reusing UDP sockets.
+explicitly allow or disallow reusing UDP sockets
 
 @item ttl=@var{ttl}
-Set the time to live value (for multicast only).
+set the time to live value (for multicast only)
 
 @item connect=@var{1|0}
 Initialize the UDP socket with @code{connect()}. In this case, the
@@ -692,28 +470,9 @@ and makes writes return with AVERROR(ECONNREFUSED) if "destination
 unreachable" is received.
 For receiving, this gives the benefit of only receiving packets from
 the specified peer address/port.
-
-@item sources=@var{address}[,@var{address}]
-Only receive packets sent to the multicast group from one of the
-specified sender IP addresses.
-
-@item block=@var{address}[,@var{address}]
-Ignore packets sent to the multicast group from the specified
-sender IP addresses.
-
-@item fifo_size=@var{units}
-Set the UDP receiving circular buffer size, expressed as a number of
-packets with size of 188 bytes. If not specified defaults to 7*4096.
-
-@item overrun_nonfatal=@var{1|0}
-Survive in case of UDP receiving circular buffer overrun. Default
-value is 0.
-
-@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
 @end table
 
-Some usage examples of the UDP protocol with @command{ffmpeg} follow.
+Some usage examples of the udp protocol with @command{ffmpeg} follow.
 
 To stream over UDP to a remote endpoint:
 @example

doc/rate_distortion.txt

@@ -23,7 +23,7 @@ Let's consider the problem of minimizing:
 
 rate is the filesize
 distortion is the quality
-lambda is a fixed value chosen as a tradeoff between quality and filesize
+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

doc/swresample.txt

@@ -31,16 +31,16 @@ Special Converter         v
                           v
                         Output
 
-Planar/Packed conversion is done when needed during sample format conversion.
-Every step can be skipped without memcpy when its not needed.
+Planar/Packed convertion is done when needed during sample format convertion
+Every step can be skiped without memcpy when its not needed.
 Either Resampling and Rematrixing can be performed first depending on which
 way its faster.
 The Buffers are needed for resampling due to resamplng being a process that
 requires future and past data, it thus also introduces inevitably a delay when
 used.
 Internally 32bit float and 16bit int is supported currently, other formats can
-easily be added.
+easily be added
 Externally all sample formats in packed and planar configuration are supported
-It's also trivial to add special converters for common cases.
-If only sample format and/or packed/planar conversion is needed, it
+Its also trivial to add special converters for common cases
+If only sample format and or packed/planar convertion is needed it
 is performed from input to output directly in a single pass with no intermediates.

doc/swscale.txt

@@ -58,7 +58,7 @@ Input to YUV Converter
 
 Horizontal scaler
     There are several horizontal scalers. A special case worth mentioning is
-    the fast bilinear scaler that is made of runtime-generated MMXEXT code
+    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.
@@ -96,3 +96,4 @@ 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, ...
+

doc/syntax.texi

@@ -1,230 +0,0 @@
-@chapter Syntax
-@c man begin SYNTAX
-
-This section documents the syntax and formats employed by the FFmpeg
-libraries and tools.
-
-@anchor{quoting_and_escaping}
-@section Quoting and escaping
-
-FFmpeg adopts the following quoting and escaping mechanism, unless
-explicitly specified. The following rules are applied:
-
-@itemize
-@item
-@code{'} and @code{\} are special characters (respectively used for
-quoting and escaping). In addition to them, there might be other
-special characters depending on the specific syntax where the escaping
-and quoting are employed.
-
-@item
-A special character is escaped by prefixing it with a '\'.
-
-@item
-All characters enclosed between '' are included literally in the
-parsed string. The quote character @code{'} itself cannot be quoted,
-so you may need to close the quote and escape it.
-
-@item
-Leading and trailing whitespaces, unless escaped or quoted, are
-removed from the parsed string.
-@end itemize
-
-Note that you may need to add a second level of escaping when using
-the command line or a script, which depends on the syntax of the
-adopted shell language.
-
-The function @code{av_get_token} defined in
-@file{libavutil/avstring.h} can be used to parse a token quoted or
-escaped according to the rules defined above.
-
-The tool @file{tools/ffescape} in the FFmpeg source tree can be used
-to automatically quote or escape a string in a script.
-
-@subsection Examples
-
-@itemize
-@item
-Escape the string @code{Crime d'Amour} containing the @code{'} special
-character:
-@example
-Crime d\'Amour
-@end example
-
-@item
-The string above contains a quote, so the @code{'} needs to be escaped
-when quoting it:
-@example
-'Crime d'\''Amour'
-@end example
-
-@item
-Include leading or trailing whitespaces using quoting:
-@example
-'  this string starts and ends with whitespaces  '
-@end example
-
-@item
-Escaping and quoting can be mixed together:
-@example
-' The string '\'string\'' is a string '
-@end example
-
-@item
-To include a literal @code{\} you can use either escaping or quoting:
-@example
-'c:\foo' can be written as c:\\foo
-@end example
-@end itemize
-
-@anchor{date syntax}
-@section Date
-
-The accepted syntax is:
-@example
-[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
-now
-@end example
-
-If the value is "now" it takes the current time.
-
-Time is local time unless 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.
-
-@anchor{time duration syntax}
-@section Time duration
-
-The accepted syntax is:
-@example
-[-]HH:MM:SS[.m...]
-[-]S+[.m...]
-@end example
-
-@var{HH} expresses the number of hours, @var{MM} the number a of minutes
-and @var{SS} the number of seconds.
-
-@anchor{video size syntax}
-@section Video size
-Specify the size of the sourced video, it may be a string of the form
-@var{width}x@var{height}, or the name of a size abbreviation.
-
-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
-
-@anchor{video rate syntax}
-@section Video rate
-
-Specify the frame rate of a video, expressed as the number of frames
-generated per second. It has to be a string in the format
-@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
-number or a valid video frame rate abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item ntsc
-30000/1001
-@item pal
-25/1
-@item qntsc
-30000/1
-@item qpal
-25/1
-@item sntsc
-30000/1
-@item spal
-25/1
-@item film
-24/1
-@item ntsc-film
-24000/1
-@end table
-
-@anchor{ratio syntax}
-@section Ratio
-
-A ratio can be expressed as an expression, or in the form
-@var{numerator}:@var{denominator}.
-
-Note that a ratio with infinite (1/0) or negative value is
-considered valid, so you should check on the returned value if you
-want to exclude those values.
-
-The undefined value can be expressed using the "0:0" string.
-
-@anchor{color syntax}
-@section Color
-
-It can be the name of a color (case insensitive match) or a
-[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
-representing the alpha component.
-
-The alpha component may be a string composed by "0x" followed by an
-hexadecimal number or a decimal number between 0.0 and 1.0, which
-represents the opacity value (0x00/0.0 means completely transparent,
-0xff/1.0 completely opaque).
-If the alpha component is not specified then 0xff is assumed.
-
-The string "random" will result in a random color.
-
-@c man end SYNTAX

doc/t2h.init

@@ -6,18 +6,73 @@ sub FFmpeg_end_section($$)
 
 $EXTRA_HEAD =
 '<link rel="icon" href="favicon.png" type="image/png" />
+<link rel="stylesheet" type="text/css" href="default.css" />
 ';
 
-$CSS_LINES = $ENV{"FFMPEG_CSS"} || <<EOT;
-<link rel="stylesheet" type="text/css" href="default.css" />
+$CSS_LINES = <<EOT;
+<style type="text/css">
+<!--
+a.summary-letter { text-decoration: none }
+a { color: #2D6198; }
+a:visited { color: #884488; }
+h1 a, h2 a, h3 a { text-decoration: inherit; color: inherit; }
+p { margin-left: 1em; margin-right: 1em; }
+table { margin-left: 2em; }
+pre { margin-left: 2em; }
+#footer { text-align: center; }
+#body { margin-left: 1em; margin-right: 1em; }
+body { background-color: #313131; margin: 0; }
+
+#container {
+    background-color: white;
+    color: #202020;
+    margin-left: 1em;
+    margin-right: 1em;
+}
+
+h1 {
+    background-color: #7BB37B;
+    border: 1px solid #6A996A;
+    color: #151515;
+    font-size: 1.2em;
+    padding-bottom: 0.2em;
+    padding-left: 0.4em;
+    padding-top: 0.2em;
+}
+
+h2 {
+    color: #313131;
+    font-size: 1.2em;
+}
+
+h3 {
+    color: #313131;
+    font-size: 0.8em;
+    margin-bottom: -8px;
+}
+
+.note {
+    margin: 1em;
+    border: 1px solid #bbc9d8;
+    background-color: #dde1e1;
+}
+
+.important {
+    margin: 1em;
+    border: 1px solid #d26767;
+    background-color: #f8e1e1;
+}
+
+-->
+</style>
 EOT
 
-my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
-<link rel="icon" href="favicon.png" type="image/png" />
-</head>
-<body>
-<div id="container">
-EOT
+my $FFMPEG_NAVBAR = $ENV{"FFMPEG_NAVBAR"} || '';
+
+$AFTER_BODY_OPEN =
+'<div id="container">' .
+"\n$FFMPEG_NAVBAR\n" .
+'<div id="body">';
 
 $PRE_BODY_CLOSE = '</div></div>';
 
@@ -28,11 +83,9 @@ $print_page_foot = \&FFmpeg_print_page_foot;
 sub FFmpeg_print_page_foot($$)
 {
     my $fh = shift;
-    my $program_string = defined &T2H_DEFAULT_program_string ?
-        T2H_DEFAULT_program_string() : program_string();
-    print $fh '<footer class="footer pagination-right">' . "\n";
-    print $fh '<span class="label label-info">' . $program_string;
-    print $fh "</span></footer></div>\n";
+    print $fh '<div id="footer">' . "\n";
+    T2H_DEFAULT_print_page_foot($fh);
+    print $fh "</div>\n";
 }
 
 $float = \&FFmpeg_float;
@@ -54,11 +107,11 @@ sub FFmpeg_float($$$$)
 
     if ($caption =~ /NOTE/)
     {
-        $class = "alert alert-info";
+        $class = "note";
     }
     elsif ($caption =~ /IMPORTANT/)
     {
-        $class = "alert alert-warning";
+        $class = "important";
     }
 
     return '<div class="float ' . $class . '">' . "$label\n" . $text . '</div>';
@@ -81,7 +134,7 @@ sub FFmpeg_print_page_head($$)
     $longtitle = "FFmpeg documentation : " . $longtitle;
 
     print $fh <<EOT;
-<!DOCTYPE html>
+$DOCTYPE
 <html>
 $Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} -->
 <!--
@@ -97,13 +150,14 @@ $description
 <meta name="Generator" content="$Texi2HTML::THISDOC{program}">
 $encoding
 $CSS_LINES
-$TEMPLATE_HEADER
+$EXTRA_HEAD
+</head>
+
+<body $BODYTEXT>
+$AFTER_BODY_OPEN
 EOT
 }
 
-# declare encoding in header
-$IN_ENCODING = $ENCODING = "utf-8";
-
 # no navigation elements
 $SECTION_NAVIGATION = 0;
 # the same for texi2html 5.0

doc/texi2pod.pl

@@ -1,4 +1,4 @@
-#! /usr/bin/perl
+#! /usr/bin/perl -w
 
 #   Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
 
@@ -23,13 +23,11 @@
 # markup to Perl POD format.  It's intended to be used to extract
 # something suitable for a manpage from a Texinfo document.
 
-use warnings;
-
 $output = 0;
 $skipping = 0;
-%chapters = ();
-@chapters_sequence = ();
-$chapter = "";
+%sects = ();
+@sects_sequence = ();
+$section = "";
 @icstack = ();
 @endwstack = ();
 @skstack = ();
@@ -38,7 +36,7 @@ $shift = "";
 %defs = ();
 $fnno = 1;
 $inf = "";
-@ibase = ();
+$ibase = "";
 
 while ($_ = shift) {
     if (/^-D(.*)$/) {
@@ -54,8 +52,6 @@ while ($_ = shift) {
         die "flags may only contain letters, digits, hyphens, dashes and underscores\n"
             unless $flag =~ /^[a-zA-Z0-9_-]+$/;
         $defs{$flag} = $value;
-    } elsif (/^-I(.*)$/) {
-        push @ibase, $1 ne "" ? $1 : shift;
     } elsif (/^-/) {
         usage();
     } else {
@@ -65,12 +61,10 @@ while ($_ = shift) {
     }
 }
 
-push @ibase, ".";
-
 if (defined $in) {
     $inf = gensym();
     open($inf, "<$in") or die "opening \"$in\": $!\n";
-    push @ibase, $1 if $in =~ m|^(.+)/[^/]+$|;
+    $ibase = $1 if $in =~ m|^(.+)/[^/]+$|;
 } else {
     $inf = \*STDIN;
 }
@@ -80,7 +74,7 @@ if (defined $out) {
 }
 
 while(defined $inf) {
-INF: while(<$inf>) {
+while(<$inf>) {
     # Certain commands are discarded without further processing.
     /^\@(?:
          [a-z]+index            # @*index: useful only in complete manual
@@ -110,28 +104,23 @@ INF: while(<$inf>) {
         push @instack, $inf;
         $inf = gensym();
 
-        for (@ibase) {
-            open($inf, "<" . $_ . "/" . $1) and next INF;
-        }
-        die "cannot open $1: $!\n";
-    };
-
-    /^\@chapter\s+([A-Za-z ]+)/ and do {
-        # close old chapter
-        $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);
-
-        # start new chapter
-        $chapter_name = $1, push (@chapters_sequence, $chapter_name) unless $skipping;
-        $chapters{$chapter_name} = "" unless exists $chapters{$chapter_name};
-        $chapter = "";
-        $output = 1;
+        # Try cwd and $ibase.
+        open($inf, "<" . $1)
+            or open($inf, "<" . $ibase . "/" . $1)
+                or die "cannot open $1 or $ibase/$1: $!\n";
         next;
     };
 
-    /^\@bye/ and do {
-        # close old chapter
-        $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);
-        last INF;
+    # 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
@@ -156,20 +145,20 @@ INF: while(<$inf>) {
         # 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 && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/;
+        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 ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) {
+        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|(?:multi|[fv])?table)$/) {
+        } elsif ($ended =~ /^(?:itemize|enumerate|[fv]?table)$/) {
             $_ = "\n=back\n";
             $ic = pop @icstack;
         } else {
@@ -196,11 +185,11 @@ INF: while(<$inf>) {
         next;
     };
 
-    /^\@(ignore|menu|iftex|ifhtml|ifnothtml)\b/ and do {
+    /^\@(ignore|menu|iftex)\b/ and do {
         push @endwstack, $endw;
         push @skstack, $skipping;
         $endw = $1;
-        $skipping = $endw !~ /ifnothtml/;
+        $skipping = 1;
         next;
     };
 
@@ -217,6 +206,7 @@ INF: while(<$inf>) {
     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())
@@ -269,16 +259,15 @@ INF: while(<$inf>) {
         $endw = "enumerate";
     };
 
-    /^\@((?:multi|[fv])?table)\s+(\@[a-z]+)/ and do {
+    /^\@([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|command)/B/;
+        $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/;
-        $ic =~ s/\@(?:columnfractions)//;
         $_ = "\n=over 4\n";
     };
 
@@ -289,21 +278,6 @@ INF: while(<$inf>) {
         $_ = "";        # need a paragraph break
     };
 
-    /^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
-        my $columns = $1;
-        $columns =~ s/\@tab/ : /;
-
-        $_ = "\n=item B&LT;". $columns ."&GT;\n";
-    };
-
-    /^\@tab\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
-        my $columns = $1;
-        $columns =~ s/\@tab/ : /;
-
-        $_ = " : ". $columns;
-        $chapter =~ s/\n+\s+$//;
-    };
-
     /^\@itemx?\s*(.+)?$/ and do {
         if (defined $1) {
             # Entity escapes prevent munging by the <> processing below.
@@ -315,7 +289,7 @@ INF: while(<$inf>) {
         }
     };
 
-    $chapter .= $shift.$_."\n";
+    $section .= $shift.$_."\n";
 }
 # End of current file.
 close($inf);
@@ -324,15 +298,16 @@ $inf = pop @instack;
 
 die "No filename or title\n" unless defined $fn && defined $tl;
 
-$chapters{NAME} = "$fn \- $tl\n";
-$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};
+$sects{NAME} = "$fn \- $tl\n";
+$sects{FOOTNOTES} .= "=back\n" if exists $sects{FOOTNOTES};
 
-unshift @chapters_sequence, "NAME";
-for $chapter (@chapters_sequence) {
-    if (exists $chapters{$chapter}) {
-        $head = uc($chapter);
+unshift @sects_sequence, "NAME";
+for $sect (@sects_sequence) {
+    if(exists $sects{$sect}) {
+        $head = $sect;
+        $head =~ s/SEEALSO/SEE ALSO/;
         print "=head1 $head\n\n";
-        print scalar unmunge ($chapters{$chapter});
+        print scalar unmunge ($sects{$sect});
         print "\n";
     }
 }
@@ -377,7 +352,6 @@ sub postprocess
     s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
     s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
     s/;\s+\@pxref\{(?:[^\}]*)\}//g;
-    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g;
     s/\@ref\{([^\}]*)\}/$1/g;
     s/\@noindent\s*//g;
     s/\@refill//g;
@@ -387,7 +361,7 @@ sub postprocess
     # @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)\{([^\},]*),?[^\}]*\}/&lt;B<$1>&gt;/g;
+    s/\@(?:uref|url|email)\{([^\},]*)\}/&lt;B<$1>&gt;/g;
     s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g;
     s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g;
 
@@ -431,13 +405,13 @@ sub unmunge
 
 sub add_footnote
 {
-    unless (exists $chapters{FOOTNOTES}) {
-        $chapters{FOOTNOTES} = "\n=over 4\n\n";
+    unless (exists $sects{FOOTNOTES}) {
+        $sects{FOOTNOTES} = "\n=over 4\n\n";
     }
 
-    $chapters{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++;
-    $chapters{FOOTNOTES} .= $_[0];
-    $chapters{FOOTNOTES} .= "\n\n";
+    $sects{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++;
+    $sects{FOOTNOTES} .= $_[0];
+    $sects{FOOTNOTES} .= "\n\n";
 }
 
 # stolen from Symbol.pm

doc/viterbi.txt

@@ -85,8 +85,8 @@ here are some edges we could choose from:
      /        \
     O-----2--4--O
 
-Finding the new best paths and scores for each point of our new column is
-trivial given we know the previous column best paths and scores:
+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
      \
@@ -107,3 +107,4 @@ one with score 3)
 
 Author: Michael niedermayer
 Copyright LGPL
+

ffmpeg.h

@@ -1,422 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_H
-#define FFMPEG_H
-
-#include "config.h"
-
-#include <stdint.h>
-#include <stdio.h>
-#include <signal.h>
-
-#if HAVE_PTHREADS
-#include <pthread.h>
-#endif
-
-#include "cmdutils.h"
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-#include "libavcodec/avcodec.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/avfiltergraph.h"
-
-#include "libavutil/avutil.h"
-#include "libavutil/dict.h"
-#include "libavutil/fifo.h"
-#include "libavutil/pixfmt.h"
-#include "libavutil/rational.h"
-
-#include "libswresample/swresample.h"
-
-#define VSYNC_AUTO       -1
-#define VSYNC_PASSTHROUGH 0
-#define VSYNC_CFR         1
-#define VSYNC_VFR         2
-#define VSYNC_DROP        0xff
-
-#define MAX_STREAMS 1024    /* arbitrary sanity check value */
-
-/* select an input stream for an output stream */
-typedef struct StreamMap {
-    int disabled;           /* 1 is this mapping is disabled by a negative map */
-    int file_index;
-    int stream_index;
-    int sync_file_index;
-    int sync_stream_index;
-    char *linklabel;       /* name of an output link, for mapping lavfi outputs */
-} StreamMap;
-
-typedef struct {
-    int  file_idx,  stream_idx,  channel_idx; // input
-    int ofile_idx, ostream_idx;               // output
-} AudioChannelMap;
-
-typedef struct OptionsContext {
-    OptionGroup *g;
-
-    /* input/output options */
-    int64_t start_time;
-    const char *format;
-
-    SpecifierOpt *codec_names;
-    int        nb_codec_names;
-    SpecifierOpt *audio_channels;
-    int        nb_audio_channels;
-    SpecifierOpt *audio_sample_rate;
-    int        nb_audio_sample_rate;
-    SpecifierOpt *frame_rates;
-    int        nb_frame_rates;
-    SpecifierOpt *frame_sizes;
-    int        nb_frame_sizes;
-    SpecifierOpt *frame_pix_fmts;
-    int        nb_frame_pix_fmts;
-
-    /* input options */
-    int64_t input_ts_offset;
-    int rate_emu;
-
-    SpecifierOpt *ts_scale;
-    int        nb_ts_scale;
-    SpecifierOpt *dump_attachment;
-    int        nb_dump_attachment;
-
-    /* output options */
-    StreamMap *stream_maps;
-    int     nb_stream_maps;
-    AudioChannelMap *audio_channel_maps; /* one info entry per -map_channel */
-    int           nb_audio_channel_maps; /* number of (valid) -map_channel settings */
-    int metadata_global_manual;
-    int metadata_streams_manual;
-    int metadata_chapters_manual;
-    const char **attachments;
-    int       nb_attachments;
-
-    int chapters_input_file;
-
-    int64_t recording_time;
-    uint64_t limit_filesize;
-    float mux_preload;
-    float mux_max_delay;
-    int shortest;
-
-    int video_disable;
-    int audio_disable;
-    int subtitle_disable;
-    int data_disable;
-
-    /* indexed by output file stream index */
-    int   *streamid_map;
-    int nb_streamid_map;
-
-    SpecifierOpt *metadata;
-    int        nb_metadata;
-    SpecifierOpt *max_frames;
-    int        nb_max_frames;
-    SpecifierOpt *bitstream_filters;
-    int        nb_bitstream_filters;
-    SpecifierOpt *codec_tags;
-    int        nb_codec_tags;
-    SpecifierOpt *sample_fmts;
-    int        nb_sample_fmts;
-    SpecifierOpt *qscale;
-    int        nb_qscale;
-    SpecifierOpt *forced_key_frames;
-    int        nb_forced_key_frames;
-    SpecifierOpt *force_fps;
-    int        nb_force_fps;
-    SpecifierOpt *frame_aspect_ratios;
-    int        nb_frame_aspect_ratios;
-    SpecifierOpt *rc_overrides;
-    int        nb_rc_overrides;
-    SpecifierOpt *intra_matrices;
-    int        nb_intra_matrices;
-    SpecifierOpt *inter_matrices;
-    int        nb_inter_matrices;
-    SpecifierOpt *top_field_first;
-    int        nb_top_field_first;
-    SpecifierOpt *metadata_map;
-    int        nb_metadata_map;
-    SpecifierOpt *presets;
-    int        nb_presets;
-    SpecifierOpt *copy_initial_nonkeyframes;
-    int        nb_copy_initial_nonkeyframes;
-    SpecifierOpt *copy_prior_start;
-    int        nb_copy_prior_start;
-    SpecifierOpt *filters;
-    int        nb_filters;
-    SpecifierOpt *reinit_filters;
-    int        nb_reinit_filters;
-    SpecifierOpt *fix_sub_duration;
-    int        nb_fix_sub_duration;
-    SpecifierOpt *pass;
-    int        nb_pass;
-    SpecifierOpt *passlogfiles;
-    int        nb_passlogfiles;
-} OptionsContext;
-
-typedef struct InputFilter {
-    AVFilterContext    *filter;
-    struct InputStream *ist;
-    struct FilterGraph *graph;
-    uint8_t            *name;
-} InputFilter;
-
-typedef struct OutputFilter {
-    AVFilterContext     *filter;
-    struct OutputStream *ost;
-    struct FilterGraph  *graph;
-    uint8_t             *name;
-
-    /* temporary storage until stream maps are processed */
-    AVFilterInOut       *out_tmp;
-} OutputFilter;
-
-typedef struct FilterGraph {
-    int            index;
-    const char    *graph_desc;
-
-    AVFilterGraph *graph;
-
-    InputFilter   **inputs;
-    int          nb_inputs;
-    OutputFilter **outputs;
-    int         nb_outputs;
-} FilterGraph;
-
-typedef struct InputStream {
-    int file_index;
-    AVStream *st;
-    int discard;             /* true if stream data should be discarded */
-    int decoding_needed;     /* true if the packets must be decoded in 'raw_fifo' */
-    AVCodec *dec;
-    AVFrame *decoded_frame;
-
-    int64_t       start;     /* time when read started */
-    /* predicted dts of the next packet read for this stream or (when there are
-     * several frames in a packet) of the next frame in current packet (in AV_TIME_BASE units) */
-    int64_t       next_dts;
-    int64_t       dts;       ///< dts of the last packet read for this stream (in AV_TIME_BASE units)
-
-    int64_t       next_pts;  ///< synthetic pts for the next decode frame (in AV_TIME_BASE units)
-    int64_t       pts;       ///< current pts of the decoded frame  (in AV_TIME_BASE units)
-    int           wrap_correction_done;
-
-    int64_t filter_in_rescale_delta_last;
-
-    double ts_scale;
-    int is_start;            /* is 1 at the start and after a discontinuity */
-    int saw_first_ts;
-    int showed_multi_packet_warning;
-    AVDictionary *opts;
-    AVRational framerate;               /* framerate forced with -r */
-    int top_field_first;
-
-    int resample_height;
-    int resample_width;
-    int resample_pix_fmt;
-
-    int      resample_sample_fmt;
-    int      resample_sample_rate;
-    int      resample_channels;
-    uint64_t resample_channel_layout;
-
-    int fix_sub_duration;
-    struct { /* previous decoded subtitle and related variables */
-        int got_output;
-        int ret;
-        AVSubtitle subtitle;
-    } prev_sub;
-
-    struct sub2video {
-        int64_t last_pts;
-        int64_t end_pts;
-        AVFilterBufferRef *ref;
-        int w, h;
-    } sub2video;
-
-    /* a pool of free buffers for decoded data */
-    FrameBuffer *buffer_pool;
-    int dr1;
-
-    /* decoded data from this stream goes into all those filters
-     * currently video and audio only */
-    InputFilter **filters;
-    int        nb_filters;
-
-    int reinit_filters;
-} InputStream;
-
-typedef struct InputFile {
-    AVFormatContext *ctx;
-    int eof_reached;      /* true if eof reached */
-    int eagain;           /* true if last read attempt returned EAGAIN */
-    int ist_index;        /* index of first stream in input_streams */
-    int64_t ts_offset;
-    int nb_streams;       /* number of stream that ffmpeg is aware of; may be different
-                             from ctx.nb_streams if new streams appear during av_read_frame() */
-    int nb_streams_warn;  /* number of streams that the user was warned of */
-    int rate_emu;
-
-#if HAVE_PTHREADS
-    pthread_t thread;           /* thread reading from this file */
-    int finished;               /* the thread has exited */
-    int joined;                 /* the thread has been joined */
-    pthread_mutex_t fifo_lock;  /* lock for access to fifo */
-    pthread_cond_t  fifo_cond;  /* the main thread will signal on this cond after reading from fifo */
-    AVFifoBuffer *fifo;         /* demuxed packets are stored here; freed by the main thread */
-#endif
-} InputFile;
-
-typedef struct OutputStream {
-    int file_index;          /* file index */
-    int index;               /* stream index in the output file */
-    int source_index;        /* InputStream index */
-    AVStream *st;            /* stream in the output file */
-    int encoding_needed;     /* true if encoding needed for this stream */
-    int frame_number;
-    /* input pts and corresponding output pts
-       for A/V sync */
-    struct InputStream *sync_ist; /* input stream to sync against */
-    int64_t sync_opts;       /* output frame counter, could be changed to some true timestamp */ // FIXME look at frame_number
-    /* pts of the first frame encoded for this stream, used for limiting
-     * recording time */
-    int64_t first_pts;
-    /* dts of the last packet sent to the muxer */
-    int64_t last_mux_dts;
-    AVBitStreamFilterContext *bitstream_filters;
-    AVCodec *enc;
-    int64_t max_frames;
-    AVFrame *filtered_frame;
-
-    /* video only */
-    AVRational frame_rate;
-    int force_fps;
-    int top_field_first;
-
-    float frame_aspect_ratio;
-
-    /* forced key frames */
-    int64_t *forced_kf_pts;
-    int forced_kf_count;
-    int forced_kf_index;
-    char *forced_keyframes;
-
-    /* audio only */
-    int audio_channels_map[SWR_CH_MAX];  /* list of the channels id to pick from the source stream */
-    int audio_channels_mapped;           /* number of channels in audio_channels_map */
-
-    char *logfile_prefix;
-    FILE *logfile;
-
-    OutputFilter *filter;
-    char *avfilter;
-
-    int64_t sws_flags;
-    int64_t swr_filter_type;
-    int64_t swr_dither_method;
-    double swr_dither_scale;
-    AVDictionary *opts;
-    int finished;        /* no more packets should be written for this stream */
-    int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
-    int stream_copy;
-    const char *attachment_filename;
-    int copy_initial_nonkeyframes;
-    int copy_prior_start;
-
-    int keep_pix_fmt;
-} OutputStream;
-
-typedef struct OutputFile {
-    AVFormatContext *ctx;
-    AVDictionary *opts;
-    int ost_index;       /* index of the first stream in output_streams */
-    int64_t recording_time;  ///< desired length of the resulting file in microseconds == AV_TIME_BASE units
-    int64_t start_time;      ///< start time in microseconds == AV_TIME_BASE units
-    uint64_t limit_filesize; /* filesize limit expressed in bytes */
-
-    int shortest;
-} OutputFile;
-
-extern InputStream **input_streams;
-extern int        nb_input_streams;
-extern InputFile   **input_files;
-extern int        nb_input_files;
-
-extern OutputStream **output_streams;
-extern int         nb_output_streams;
-extern OutputFile   **output_files;
-extern int         nb_output_files;
-
-extern FilterGraph **filtergraphs;
-extern int        nb_filtergraphs;
-
-extern char *vstats_filename;
-
-extern float audio_drift_threshold;
-extern float dts_delta_threshold;
-extern float dts_error_threshold;
-
-extern int audio_volume;
-extern int audio_sync_method;
-extern int video_sync_method;
-extern int do_benchmark;
-extern int do_benchmark_all;
-extern int do_deinterlace;
-extern int do_hex_dump;
-extern int do_pkt_dump;
-extern int copy_ts;
-extern int copy_tb;
-extern int debug_ts;
-extern int exit_on_error;
-extern int print_stats;
-extern int qp_hist;
-extern int stdin_interaction;
-extern int frame_bits_per_raw_sample;
-extern AVIOContext *progress_avio;
-
-extern const AVIOInterruptCB int_cb;
-
-extern const OptionDef options[];
-
-void term_init(void);
-void term_exit(void);
-
-void reset_options(OptionsContext *o, int is_input);
-void show_usage(void);
-
-void opt_output_file(void *optctx, const char *filename);
-
-void assert_avoptions(AVDictionary *m);
-
-int guess_input_channel_layout(InputStream *ist);
-
-enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFormat target);
-void choose_sample_fmt(AVStream *st, AVCodec *codec);
-
-int configure_filtergraph(FilterGraph *fg);
-int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out);
-int ist_in_filtergraph(FilterGraph *fg, InputStream *ist);
-FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
-
-int ffmpeg_parse_options(int argc, char **argv);
-
-#endif /* FFMPEG_H */

ffmpeg_filter.c

@@ -1,793 +0,0 @@
-/*
- * ffmpeg filter configuration
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg 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
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "ffmpeg.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/avfiltergraph.h"
-#include "libavfilter/buffersink.h"
-
-#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
-#include "libavutil/bprint.h"
-#include "libavutil/channel_layout.h"
-#include "libavutil/opt.h"
-#include "libavutil/pixdesc.h"
-#include "libavutil/pixfmt.h"
-#include "libavutil/imgutils.h"
-#include "libavutil/samplefmt.h"
-
-enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFormat target)
-{
-    if (codec && codec->pix_fmts) {
-        const enum AVPixelFormat *p = codec->pix_fmts;
-        const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
-        int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
-        enum AVPixelFormat best= AV_PIX_FMT_NONE;
-        const enum AVPixelFormat mjpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
-        const enum AVPixelFormat ljpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
-                                                     AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
-
-        if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
-            if (st->codec->codec_id == AV_CODEC_ID_MJPEG) {
-                p = mjpeg_formats;
-            } else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) {
-                p =ljpeg_formats;
-            }
-        }
-        for (; *p != AV_PIX_FMT_NONE; p++) {
-            best= avcodec_find_best_pix_fmt_of_2(best, *p, target, has_alpha, NULL);
-            if (*p == target)
-                break;
-        }
-        if (*p == AV_PIX_FMT_NONE) {
-            if (target != AV_PIX_FMT_NONE)
-                av_log(NULL, AV_LOG_WARNING,
-                       "Incompatible pixel format '%s' for codec '%s', auto-selecting format '%s'\n",
-                       av_get_pix_fmt_name(target),
-                       codec->name,
-                       av_get_pix_fmt_name(best));
-            return best;
-        }
-    }
-    return target;
-}
-
-void choose_sample_fmt(AVStream *st, AVCodec *codec)
-{
-    if (codec && codec->sample_fmts) {
-        const enum AVSampleFormat *p = codec->sample_fmts;
-        for (; *p != -1; p++) {
-            if (*p == st->codec->sample_fmt)
-                break;
-        }
-        if (*p == -1) {
-            if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
-                av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
-            if(av_get_sample_fmt_name(st->codec->sample_fmt))
-            av_log(NULL, AV_LOG_WARNING,
-                   "Incompatible sample format '%s' for codec '%s', auto-selecting format '%s'\n",
-                   av_get_sample_fmt_name(st->codec->sample_fmt),
-                   codec->name,
-                   av_get_sample_fmt_name(codec->sample_fmts[0]));
-            st->codec->sample_fmt = codec->sample_fmts[0];
-        }
-    }
-}
-
-static char *choose_pix_fmts(OutputStream *ost)
-{
-     if (ost->keep_pix_fmt) {
-        if (ost->filter)
-            avfilter_graph_set_auto_convert(ost->filter->graph->graph,
-                                            AVFILTER_AUTO_CONVERT_NONE);
-        if (ost->st->codec->pix_fmt == AV_PIX_FMT_NONE)
-            return NULL;
-        return av_strdup(av_get_pix_fmt_name(ost->st->codec->pix_fmt));
-    }
-    if (ost->st->codec->pix_fmt != AV_PIX_FMT_NONE) {
-        return av_strdup(av_get_pix_fmt_name(choose_pixel_fmt(ost->st, ost->enc, ost->st->codec->pix_fmt)));
-    } else if (ost->enc && ost->enc->pix_fmts) {
-        const enum AVPixelFormat *p;
-        AVIOContext *s = NULL;
-        uint8_t *ret;
-        int len;
-
-        if (avio_open_dyn_buf(&s) < 0)
-            exit(1);
-
-        p = ost->enc->pix_fmts;
-        if (ost->st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
-            if (ost->st->codec->codec_id == AV_CODEC_ID_MJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
-            } else if (ost->st->codec->codec_id == AV_CODEC_ID_LJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
-                                                    AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
-            }
-        }
-
-        for (; *p != AV_PIX_FMT_NONE; p++) {
-            const char *name = av_get_pix_fmt_name(*p);
-            avio_printf(s, "%s:", name);
-        }
-        len = avio_close_dyn_buf(s, &ret);
-        ret[len - 1] = 0;
-        return ret;
-    } else
-        return NULL;
-}
-
-/* Define a function for building a string containing a list of
- * allowed formats. */
-#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name, separator)\
-static char *choose_ ## var ## s(OutputStream *ost)                            \
-{                                                                              \
-    if (ost->st->codec->var != none) {                                         \
-        get_name(ost->st->codec->var);                                         \
-        return av_strdup(name);                                                \
-    } else if (ost->enc->supported_list) {                                     \
-        const type *p;                                                         \
-        AVIOContext *s = NULL;                                                 \
-        uint8_t *ret;                                                          \
-        int len;                                                               \
-                                                                               \
-        if (avio_open_dyn_buf(&s) < 0)                                         \
-            exit(1);                                                           \
-                                                                               \
-        for (p = ost->enc->supported_list; *p != none; p++) {                  \
-            get_name(*p);                                                      \
-            avio_printf(s, "%s" separator, name);                              \
-        }                                                                      \
-        len = avio_close_dyn_buf(s, &ret);                                     \
-        ret[len - 1] = 0;                                                      \
-        return ret;                                                            \
-    } else                                                                     \
-        return NULL;                                                           \
-}
-
-// DEF_CHOOSE_FORMAT(enum AVPixelFormat, pix_fmt, pix_fmts, AV_PIX_FMT_NONE,
-//                   GET_PIX_FMT_NAME, ":")
-
-DEF_CHOOSE_FORMAT(enum AVSampleFormat, sample_fmt, sample_fmts,
-                  AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME, ",")
-
-DEF_CHOOSE_FORMAT(int, sample_rate, supported_samplerates, 0,
-                  GET_SAMPLE_RATE_NAME, ",")
-
-DEF_CHOOSE_FORMAT(uint64_t, channel_layout, channel_layouts, 0,
-                  GET_CH_LAYOUT_NAME, ",")
-
-FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost)
-{
-    FilterGraph *fg = av_mallocz(sizeof(*fg));
-
-    if (!fg)
-        exit(1);
-    fg->index = nb_filtergraphs;
-
-    GROW_ARRAY(fg->outputs, fg->nb_outputs);
-    if (!(fg->outputs[0] = av_mallocz(sizeof(*fg->outputs[0]))))
-        exit(1);
-    fg->outputs[0]->ost   = ost;
-    fg->outputs[0]->graph = fg;
-
-    ost->filter = fg->outputs[0];
-
-    GROW_ARRAY(fg->inputs, fg->nb_inputs);
-    if (!(fg->inputs[0] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit(1);
-    fg->inputs[0]->ist   = ist;
-    fg->inputs[0]->graph = fg;
-
-    GROW_ARRAY(ist->filters, ist->nb_filters);
-    ist->filters[ist->nb_filters - 1] = fg->inputs[0];
-
-    GROW_ARRAY(filtergraphs, nb_filtergraphs);
-    filtergraphs[nb_filtergraphs - 1] = fg;
-
-    return fg;
-}
-
-static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
-{
-    InputStream *ist = NULL;
-    enum AVMediaType type = avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx);
-    int i;
-
-    // TODO: support other filter types
-    if (type != AVMEDIA_TYPE_VIDEO && type != AVMEDIA_TYPE_AUDIO) {
-        av_log(NULL, AV_LOG_FATAL, "Only video and audio filters supported "
-               "currently.\n");
-        exit(1);
-    }
-
-    if (in->name) {
-        AVFormatContext *s;
-        AVStream       *st = NULL;
-        char *p;
-        int file_idx = strtol(in->name, &p, 0);
-
-        if (file_idx < 0 || file_idx >= nb_input_files) {
-            av_log(NULL, AV_LOG_FATAL, "Invalid file index %d in filtergraph description %s.\n",
-                   file_idx, fg->graph_desc);
-            exit(1);
-        }
-        s = input_files[file_idx]->ctx;
-
-        for (i = 0; i < s->nb_streams; i++) {
-            enum AVMediaType stream_type = s->streams[i]->codec->codec_type;
-            if (stream_type != type &&
-                !(stream_type == AVMEDIA_TYPE_SUBTITLE &&
-                  type == AVMEDIA_TYPE_VIDEO /* sub2video hack */))
-                continue;
-            if (check_stream_specifier(s, s->streams[i], *p == ':' ? p + 1 : p) == 1) {
-                st = s->streams[i];
-                break;
-            }
-        }
-        if (!st) {
-            av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s "
-                   "matches no streams.\n", p, fg->graph_desc);
-            exit(1);
-        }
-        ist = input_streams[input_files[file_idx]->ist_index + st->index];
-    } else {
-        /* find the first unused stream of corresponding type */
-        for (i = 0; i < nb_input_streams; i++) {
-            ist = input_streams[i];
-            if (ist->st->codec->codec_type == type && ist->discard)
-                break;
-        }
-        if (i == nb_input_streams) {
-            av_log(NULL, AV_LOG_FATAL, "Cannot find a matching stream for "
-                   "unlabeled input pad %d on filter %s\n", in->pad_idx,
-                   in->filter_ctx->name);
-            exit(1);
-        }
-    }
-    av_assert0(ist);
-
-    ist->discard         = 0;
-    ist->decoding_needed++;
-    ist->st->discard = AVDISCARD_NONE;
-
-    GROW_ARRAY(fg->inputs, fg->nb_inputs);
-    if (!(fg->inputs[fg->nb_inputs - 1] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit(1);
-    fg->inputs[fg->nb_inputs - 1]->ist   = ist;
-    fg->inputs[fg->nb_inputs - 1]->graph = fg;
-
-    GROW_ARRAY(ist->filters, ist->nb_filters);
-    ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
-}
-
-static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
-{
-    char *pix_fmts;
-    OutputStream *ost = ofilter->ost;
-    AVCodecContext *codec = ost->st->codec;
-    AVFilterContext *last_filter = out->filter_ctx;
-    int pad_idx = out->pad_idx;
-    int ret;
-    char name[255];
-    AVBufferSinkParams *buffersink_params = av_buffersink_params_alloc();
-
-    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
-    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("ffbuffersink"),
-                                       name, NULL, NULL, fg->graph);
-    av_freep(&buffersink_params);
-
-    if (ret < 0)
-        return ret;
-
-    if (codec->width || codec->height) {
-        char args[255];
-        AVFilterContext *filter;
-
-        snprintf(args, sizeof(args), "%d:%d:flags=0x%X",
-                 codec->width,
-                 codec->height,
-                 (unsigned)ost->sws_flags);
-        snprintf(name, sizeof(name), "scaler for output stream %d:%d",
-                 ost->file_index, ost->index);
-        if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
-                                                name, args, NULL, fg->graph)) < 0)
-            return ret;
-        if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
-            return ret;
-
-        last_filter = filter;
-        pad_idx = 0;
-    }
-
-    if ((pix_fmts = choose_pix_fmts(ost))) {
-        AVFilterContext *filter;
-        snprintf(name, sizeof(name), "pixel format for output stream %d:%d",
-                 ost->file_index, ost->index);
-        if ((ret = avfilter_graph_create_filter(&filter,
-                                                avfilter_get_by_name("format"),
-                                                "format", pix_fmts, NULL,
-                                                fg->graph)) < 0)
-            return ret;
-        if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
-            return ret;
-
-        last_filter = filter;
-        pad_idx     = 0;
-        av_freep(&pix_fmts);
-    }
-
-    if (ost->frame_rate.num && 0) {
-        AVFilterContext *fps;
-        char args[255];
-
-        snprintf(args, sizeof(args), "fps=%d/%d", ost->frame_rate.num,
-                 ost->frame_rate.den);
-        snprintf(name, sizeof(name), "fps for output stream %d:%d",
-                 ost->file_index, ost->index);
-        ret = avfilter_graph_create_filter(&fps, avfilter_get_by_name("fps"),
-                                           name, args, NULL, fg->graph);
-        if (ret < 0)
-            return ret;
-
-        ret = avfilter_link(last_filter, pad_idx, fps, 0);
-        if (ret < 0)
-            return ret;
-        last_filter = fps;
-        pad_idx = 0;
-    }
-
-    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
-        return ret;
-
-    return 0;
-}
-
-static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
-{
-    OutputStream *ost = ofilter->ost;
-    AVCodecContext *codec  = ost->st->codec;
-    AVFilterContext *last_filter = out->filter_ctx;
-    int pad_idx = out->pad_idx;
-    char *sample_fmts, *sample_rates, *channel_layouts;
-    char name[255];
-    int ret;
-
-
-    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
-    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("ffabuffersink"),
-                                       name, NULL, NULL, fg->graph);
-    if (ret < 0)
-        return ret;
-
-#define AUTO_INSERT_FILTER(opt_name, filter_name, arg) do {                 \
-    AVFilterContext *filt_ctx;                                              \
-                                                                            \
-    av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi "            \
-           "similarly to -af " filter_name "=%s.\n", arg);                  \
-                                                                            \
-    ret = avfilter_graph_create_filter(&filt_ctx,                           \
-                                       avfilter_get_by_name(filter_name),   \
-                                       filter_name, arg, NULL, fg->graph);  \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    ret = avfilter_link(last_filter, pad_idx, filt_ctx, 0);                 \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    last_filter = filt_ctx;                                                 \
-    pad_idx = 0;                                                            \
-} while (0)
-    if (ost->audio_channels_mapped) {
-        int i;
-        AVBPrint pan_buf;
-        av_bprint_init(&pan_buf, 256, 8192);
-        av_bprintf(&pan_buf, "0x%"PRIx64,
-                   av_get_default_channel_layout(ost->audio_channels_mapped));
-        for (i = 0; i < ost->audio_channels_mapped; i++)
-            if (ost->audio_channels_map[i] != -1)
-                av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
-
-        AUTO_INSERT_FILTER("-map_channel", "pan", pan_buf.str);
-        av_bprint_finalize(&pan_buf, NULL);
-    }
-
-    if (codec->channels && !codec->channel_layout)
-        codec->channel_layout = av_get_default_channel_layout(codec->channels);
-
-    sample_fmts     = choose_sample_fmts(ost);
-    sample_rates    = choose_sample_rates(ost);
-    channel_layouts = choose_channel_layouts(ost);
-    if (sample_fmts || sample_rates || channel_layouts) {
-        AVFilterContext *format;
-        char args[256];
-        args[0] = 0;
-
-        if (sample_fmts)
-            av_strlcatf(args, sizeof(args), "sample_fmts=%s:",
-                            sample_fmts);
-        if (sample_rates)
-            av_strlcatf(args, sizeof(args), "sample_rates=%s:",
-                            sample_rates);
-        if (channel_layouts)
-            av_strlcatf(args, sizeof(args), "channel_layouts=%s:",
-                            channel_layouts);
-
-        av_freep(&sample_fmts);
-        av_freep(&sample_rates);
-        av_freep(&channel_layouts);
-
-        snprintf(name, sizeof(name), "audio format for output stream %d:%d",
-                 ost->file_index, ost->index);
-        ret = avfilter_graph_create_filter(&format,
-                                           avfilter_get_by_name("aformat"),
-                                           name, args, NULL, fg->graph);
-        if (ret < 0)
-            return ret;
-
-        ret = avfilter_link(last_filter, pad_idx, format, 0);
-        if (ret < 0)
-            return ret;
-
-        last_filter = format;
-        pad_idx = 0;
-    }
-
-    if (audio_volume != 256 && 0) {
-        char args[256];
-
-        snprintf(args, sizeof(args), "%f", audio_volume / 256.);
-        AUTO_INSERT_FILTER("-vol", "volume", args);
-    }
-
-    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
-        return ret;
-
-    return 0;
-}
-
-#define DESCRIBE_FILTER_LINK(f, inout, in)                         \
-{                                                                  \
-    AVFilterContext *ctx = inout->filter_ctx;                      \
-    AVFilterPad *pads = in ? ctx->input_pads  : ctx->output_pads;  \
-    int       nb_pads = in ? ctx->input_count : ctx->output_count; \
-    AVIOContext *pb;                                               \
-                                                                   \
-    if (avio_open_dyn_buf(&pb) < 0)                                \
-        exit(1);                                                   \
-                                                                   \
-    avio_printf(pb, "%s", ctx->filter->name);                      \
-    if (nb_pads > 1)                                               \
-        avio_printf(pb, ":%s", avfilter_pad_get_name(pads, inout->pad_idx));\
-    avio_w8(pb, 0);                                                \
-    avio_close_dyn_buf(pb, &f->name);                              \
-}
-
-int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
-{
-    av_freep(&ofilter->name);
-    DESCRIBE_FILTER_LINK(ofilter, out, 0);
-
-    switch (avfilter_pad_get_type(out->filter_ctx->output_pads, out->pad_idx)) {
-    case AVMEDIA_TYPE_VIDEO: return configure_output_video_filter(fg, ofilter, out);
-    case AVMEDIA_TYPE_AUDIO: return configure_output_audio_filter(fg, ofilter, out);
-    default: av_assert0(0);
-    }
-}
-
-static int sub2video_prepare(InputStream *ist)
-{
-    AVFormatContext *avf = input_files[ist->file_index]->ctx;
-    int i, ret, w, h;
-    uint8_t *image[4];
-    int linesize[4];
-
-    /* Compute the size of the canvas for the subtitles stream.
-       If the subtitles codec has set a size, use it. Otherwise use the
-       maximum dimensions of the video streams in the same file. */
-    w = ist->st->codec->width;
-    h = ist->st->codec->height;
-    if (!(w && h)) {
-        for (i = 0; i < avf->nb_streams; i++) {
-            if (avf->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
-                w = FFMAX(w, avf->streams[i]->codec->width);
-                h = FFMAX(h, avf->streams[i]->codec->height);
-            }
-        }
-        if (!(w && h)) {
-            w = FFMAX(w, 720);
-            h = FFMAX(h, 576);
-        }
-        av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h);
-    }
-    ist->sub2video.w = ist->st->codec->width  = ist->resample_width  = w;
-    ist->sub2video.h = ist->st->codec->height = ist->resample_height = h;
-
-    /* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the
-       palettes for all rectangles are identical or compatible */
-    ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32;
-
-    ret = av_image_alloc(image, linesize, w, h, AV_PIX_FMT_RGB32, 32);
-    if (ret < 0)
-        return ret;
-    memset(image[0], 0, h * linesize[0]);
-    ist->sub2video.ref = avfilter_get_video_buffer_ref_from_arrays(
-            image, linesize, AV_PERM_READ | AV_PERM_PRESERVE,
-            w, h, AV_PIX_FMT_RGB32);
-    if (!ist->sub2video.ref) {
-        av_free(image[0]);
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
-                                        AVFilterInOut *in)
-{
-    AVFilterContext *first_filter = in->filter_ctx;
-    AVFilter *filter = avfilter_get_by_name("buffer");
-    InputStream *ist = ifilter->ist;
-    AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) :
-                                         ist->st->time_base;
-    AVRational fr = ist->framerate.num ? ist->framerate :
-                                         ist->st->r_frame_rate;
-    AVRational sar;
-    AVBPrint args;
-    char name[255];
-    int pad_idx = in->pad_idx;
-    int ret;
-
-    if (!ist->framerate.num && ist->st->codec->ticks_per_frame>1) {
-        AVRational codec_fr = av_inv_q(ist->st->codec->time_base);
-        AVRational   avg_fr = ist->st->avg_frame_rate;
-        codec_fr.den *= ist->st->codec->ticks_per_frame;
-        if (   codec_fr.num>0 && codec_fr.den>0 && av_q2d(codec_fr) < av_q2d(fr)*0.7
-            && fabs(1.0 - av_q2d(av_div_q(avg_fr, fr)))>0.1)
-            fr = codec_fr;
-    }
-
-    if (ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
-        ret = sub2video_prepare(ist);
-        if (ret < 0)
-            return ret;
-    }
-
-    sar = ist->st->sample_aspect_ratio.num ?
-          ist->st->sample_aspect_ratio :
-          ist->st->codec->sample_aspect_ratio;
-    if(!sar.den)
-        sar = (AVRational){0,1};
-    av_bprint_init(&args, 0, 1);
-    av_bprintf(&args,
-             "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
-             "pixel_aspect=%d/%d:sws_param=flags=%d", ist->resample_width,
-             ist->resample_height, ist->resample_pix_fmt,
-             tb.num, tb.den, sar.num, sar.den,
-             SWS_BILINEAR + ((ist->st->codec->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
-    if (fr.num && fr.den)
-        av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
-    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
-             ist->file_index, ist->st->index);
-
-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter, name,
-                                            args.str, NULL, fg->graph)) < 0)
-        return ret;
-
-    if (ist->framerate.num) {
-        AVFilterContext *setpts;
-
-        snprintf(name, sizeof(name), "force CFR for input from stream %d:%d",
-                 ist->file_index, ist->st->index);
-        if ((ret = avfilter_graph_create_filter(&setpts,
-                                                avfilter_get_by_name("setpts"),
-                                                name, "N", NULL,
-                                                fg->graph)) < 0)
-            return ret;
-
-        if ((ret = avfilter_link(setpts, 0, first_filter, pad_idx)) < 0)
-            return ret;
-
-        first_filter = setpts;
-        pad_idx = 0;
-    }
-
-    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
-        return ret;
-    return 0;
-}
-
-static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
-                                        AVFilterInOut *in)
-{
-    AVFilterContext *first_filter = in->filter_ctx;
-    AVFilter *filter = avfilter_get_by_name("abuffer");
-    InputStream *ist = ifilter->ist;
-    int pad_idx = in->pad_idx;
-    char args[255], name[255];
-    int ret;
-
-    snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s"
-             ":channel_layout=0x%"PRIx64,
-             1, ist->st->codec->sample_rate,
-             ist->st->codec->sample_rate,
-             av_get_sample_fmt_name(ist->st->codec->sample_fmt),
-             ist->st->codec->channel_layout);
-    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
-             ist->file_index, ist->st->index);
-
-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter,
-                                            name, args, NULL,
-                                            fg->graph)) < 0)
-        return ret;
-
-#define AUTO_INSERT_FILTER_INPUT(opt_name, filter_name, arg) do {                 \
-    AVFilterContext *filt_ctx;                                              \
-                                                                            \
-    av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi "            \
-           "similarly to -af " filter_name "=%s.\n", arg);                  \
-                                                                            \
-    snprintf(name, sizeof(name), "graph %d %s for input stream %d:%d",      \
-                fg->index, filter_name, ist->file_index, ist->st->index);   \
-    ret = avfilter_graph_create_filter(&filt_ctx,                           \
-                                       avfilter_get_by_name(filter_name),   \
-                                       name, arg, NULL, fg->graph);         \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    ret = avfilter_link(filt_ctx, 0, first_filter, pad_idx);                \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    first_filter = filt_ctx;                                                  \
-} while (0)
-
-    if (audio_sync_method > 0) {
-        char args[256] = {0};
-
-        av_strlcatf(args, sizeof(args), "async=%d", audio_sync_method);
-        if (audio_drift_threshold != 0.1)
-            av_strlcatf(args, sizeof(args), ":min_hard_comp=%f", audio_drift_threshold);
-        AUTO_INSERT_FILTER_INPUT("-async", "aresample", args);
-    }
-
-//     if (ost->audio_channels_mapped) {
-//         int i;
-//         AVBPrint pan_buf;
-//         av_bprint_init(&pan_buf, 256, 8192);
-//         av_bprintf(&pan_buf, "0x%"PRIx64,
-//                    av_get_default_channel_layout(ost->audio_channels_mapped));
-//         for (i = 0; i < ost->audio_channels_mapped; i++)
-//             if (ost->audio_channels_map[i] != -1)
-//                 av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
-//         AUTO_INSERT_FILTER_INPUT("-map_channel", "pan", pan_buf.str);
-//         av_bprint_finalize(&pan_buf, NULL);
-//     }
-
-    if (audio_volume != 256) {
-        char args[256];
-
-        av_log(NULL, AV_LOG_WARNING, "-vol has been deprecated. Use the volume "
-               "audio filter instead.\n");
-
-        snprintf(args, sizeof(args), "%f", audio_volume / 256.);
-        AUTO_INSERT_FILTER_INPUT("-vol", "volume", args);
-    }
-    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
-        return ret;
-
-    return 0;
-}
-
-static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
-                                  AVFilterInOut *in)
-{
-    av_freep(&ifilter->name);
-    DESCRIBE_FILTER_LINK(ifilter, in, 1);
-
-    switch (avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx)) {
-    case AVMEDIA_TYPE_VIDEO: return configure_input_video_filter(fg, ifilter, in);
-    case AVMEDIA_TYPE_AUDIO: return configure_input_audio_filter(fg, ifilter, in);
-    default: av_assert0(0);
-    }
-}
-
-int configure_filtergraph(FilterGraph *fg)
-{
-    AVFilterInOut *inputs, *outputs, *cur;
-    int ret, i, init = !fg->graph, simple = !fg->graph_desc;
-    const char *graph_desc = simple ? fg->outputs[0]->ost->avfilter :
-                                      fg->graph_desc;
-
-    avfilter_graph_free(&fg->graph);
-    if (!(fg->graph = avfilter_graph_alloc()))
-        return AVERROR(ENOMEM);
-
-    if (simple) {
-        OutputStream *ost = fg->outputs[0]->ost;
-        char args[255];
-        snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
-        fg->graph->scale_sws_opts = av_strdup(args);
-
-        args[0] = 0;
-        if (ost->swr_filter_type != SWR_FILTER_TYPE_KAISER)
-            av_strlcatf(args, sizeof(args), "filter_type=%d:", (int)ost->swr_filter_type);
-        if (ost->swr_dither_method)
-            av_strlcatf(args, sizeof(args), "dither_method=%d:", (int)ost->swr_dither_method);
-        if (ost->swr_dither_scale != 1.0)
-            av_strlcatf(args, sizeof(args), "dither_scale=%f:", ost->swr_dither_scale);
-        if (strlen(args))
-            args[strlen(args)-1] = 0;
-        av_opt_set(fg->graph, "aresample_swr_opts", args, 0);
-    }
-
-    if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
-        return ret;
-
-    if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
-        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' does not have "
-               "exactly one input and output.\n", graph_desc);
-        return AVERROR(EINVAL);
-    }
-
-    for (cur = inputs; !simple && init && cur; cur = cur->next)
-        init_input_filter(fg, cur);
-
-    for (cur = inputs, i = 0; cur; cur = cur->next, i++)
-        if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0)
-            return ret;
-    avfilter_inout_free(&inputs);
-
-    if (!init || simple) {
-        /* we already know the mappings between lavfi outputs and output streams,
-         * so we can finish the setup */
-        for (cur = outputs, i = 0; cur; cur = cur->next, i++)
-            configure_output_filter(fg, fg->outputs[i], cur);
-        avfilter_inout_free(&outputs);
-
-        if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
-            return ret;
-    } else {
-        /* wait until output mappings are processed */
-        for (cur = outputs; cur;) {
-            GROW_ARRAY(fg->outputs, fg->nb_outputs);
-            if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
-                exit(1);
-            fg->outputs[fg->nb_outputs - 1]->graph   = fg;
-            fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
-            cur = cur->next;
-            fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
-        }
-    }
-
-    return 0;
-}
-
-int ist_in_filtergraph(FilterGraph *fg, InputStream *ist)
-{
-    int i;
-    for (i = 0; i < fg->nb_inputs; i++)
-        if (fg->inputs[i]->ist == ist)
-            return 1;
-    return 0;
-}
-