H.264: fix 4:4:4 + deblocking + 8x8dct + cavlc + MBAFF

(cherry picked from commit 2702a6f114)
H.264: fix 4:4:4 + deblocking + MBAFF
2011-06-23 00:17:03 +02:00 · 2011-06-23 00:17:03 +02:00 · 2011-06-23 00:17:03 +02:00 · 2011-06-23 00:17:03 +02:00 · 2011-06-22 23:21:19 +02:00 · 2011-06-22 23:19:34 +02:00
3399 changed files with 189262 additions and 402448 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -1,68 +1,48 @@
-*.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
+config.*
+doc/*.1
+doc/*.html
+doc/*.pod
+doxy
+ffmpeg
+ffplay
+ffprobe
+ffserver
+libavcodec/*_tablegen
+libavcodec/*_tables.c
+libavcodec/*_tables.h
+libavcodec/libavcodec*
+libavcore/libavcore*
+libavdevice/libavdevice*
+libavfilter/libavfilter*
+libavformat/libavformat*
+libavutil/avconfig.h
+libavutil/libavutil*
+libpostproc/libpostproc*
+libswscale/libswscale*
+tests/audiogen
+tests/base64
+tests/data
+tests/rotozoom
+tests/seek_test
+tests/tiny_psnr
+tests/videogen
+tests/vsynth1
+tests/vsynth2
+tools/cws2fws
+tools/graph2dot
+tools/lavfi-showfiltfmts
+tools/pktdumper
+tools/probetest
+tools/qt-faststart
+tools/trasher
+tools/trasher*.d
+version.h
--- a/COPYING.LGPLv2.1
+++ b/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!
+
+
--- a/706
+++ b/706
@@ -0,0 +1,706 @@
+Entries are sorted chronologically from oldest to youngest within each release,
+releases are sorted from youngest to oldest.
+
+
+version 0.7:
+
+- many many things we forgot because we rather write code than changelogs
+- 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
+
+
+version 0.7_beta2:
+
+- 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
+- DPX image encoder
+- 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
+
+
+version 0.7_beta1:
+
+- 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
+- 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
+- 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 asm functions
+- MJPEG/AVI1 to JPEG/JFIF bitstream filter
+- ASS subtitle encoder and decoder
+- IEC 61937 encapsulation for E-AC3, 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
+
+
+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
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -1,4 +1,4 @@
-# Doxyfile 1.7.1
+# Doxyfile 1.5.6

 # This file describes the settings to be used by the documentation system
 # doxygen (www.doxygen.org) for a project
@@ -31,20 +31,14 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         = 1.1.14
-
-# 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
-# pixels and the maximum width should not exceed 200 pixels. Doxygen will
-# copy the logo to the output directory.
-PROJECT_LOGO           =
+PROJECT_NUMBER         =

 # The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute)
 # base path where the generated documentation will be put.
 # 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
@@ -60,11 +54,11 @@ CREATE_SUBDIRS         = NO
 # information to generate all constant output in the proper language.
 # The default language is English, other supported languages are:
 # Afrikaans, Arabic, Brazilian, Catalan, Chinese, Chinese-Traditional,
-# Croatian, Czech, Danish, Dutch, Esperanto, Farsi, Finnish, French, German,
-# Greek, Hungarian, Italian, Japanese, Japanese-en (Japanese with English
-# messages), Korean, Korean-en, Lithuanian, Norwegian, Macedonian, Persian,
-# Polish, Portuguese, Romanian, Russian, Serbian, Serbian-Cyrilic, Slovak,
-# Slovene, Spanish, Swedish, Ukrainian, and Vietnamese.
+# Croatian, Czech, Danish, Dutch, Farsi, Finnish, French, German, Greek,
+# Hungarian, Italian, Japanese, Japanese-en (Japanese with English messages),
+# Korean, Korean-en, Lithuanian, Norwegian, Macedonian, Persian, Polish,
+# Portuguese, Romanian, Russian, Serbian, Slovak, Slovene, Spanish, Swedish,
+# and Ukrainian.

 OUTPUT_LANGUAGE        = English

@@ -161,6 +155,13 @@ QT_AUTOBRIEF           = NO

 MULTILINE_CPP_IS_BRIEF = NO

+# If the DETAILS_AT_TOP tag is set to YES then Doxygen
+# will output the detailed description near the top, like JavaDoc.
+# If set to NO, the detailed description appears after the member
+# documentation.
+
+DETAILS_AT_TOP         = NO
+
 # If the INHERIT_DOCS tag is set to YES (the default) then an undocumented
 # member inherits the documentation from any documented member that it
 # re-implements.
@@ -213,18 +214,6 @@ OPTIMIZE_FOR_FORTRAN   = NO

 OPTIMIZE_OUTPUT_VHDL   = NO

-# Doxygen selects the parser to use depending on the extension of the files it
-# parses. With this tag you can assign which parser to use for a given extension.
-# Doxygen has a built-in mapping, but you can override or extend it using this
-# tag. The format is ext=language, where ext is a file extension, and language
-# is one of the parsers supported by doxygen: IDL, Java, Javascript, CSharp, C,
-# C++, D, PHP, Objective-C, Python, Fortran, VHDL, C, C++. For instance to make
-# doxygen treat .inc files as Fortran files (default is PHP), and .f files as C
-# (default is Fortran), use: inc=Fortran f=C. Note that for custom extensions
-# you also need to set FILE_PATTERNS otherwise the files are not read by doxygen.
-
-EXTENSION_MAPPING      =
-
 # If you use STL classes (i.e. std::string, std::vector, etc.) but do not want
 # to include (a tag file for) the STL sources as input, then you should
 # set this tag to YES in order to let doxygen match functions declarations and
@@ -279,22 +268,6 @@ SUBGROUPING            = YES

 TYPEDEF_HIDES_STRUCT   = NO

-# The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
-# determine which symbols to keep in memory and which to flush to disk.
-# When the cache is full, less often used symbols will be written to disk.
-# For small to medium size projects (<1000 input files) the default value is
-# probably good enough. For larger projects a too small cache size can cause
-# doxygen to be busy swapping symbols to and from disk most of the time
-# 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
-# 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
-
-SYMBOL_CACHE_SIZE      = 0
-
 #---------------------------------------------------------------------------
 # Build related configuration options
 #---------------------------------------------------------------------------
@@ -393,12 +366,6 @@ HIDE_SCOPE_NAMES       = NO

 SHOW_INCLUDE_FILES     = YES

-# If the FORCE_LOCAL_INCLUDES tag is set to YES then Doxygen
-# will list include files with double quotes in the documentation
-# rather than with sharp brackets.
-
-FORCE_LOCAL_INCLUDES   = NO
-
 # If the INLINE_INFO tag is set to YES (the default) then a tag [inline]
 # is inserted in the documentation for inline members.

@@ -418,16 +385,6 @@ SORT_MEMBER_DOCS       = YES

 SORT_BRIEF_DOCS        = NO

-# If the SORT_MEMBERS_CTORS_1ST tag is set to YES then doxygen
-# will sort the (brief and detailed) documentation of class members so that
-# constructors and destructors are listed first. If set to NO (the default)
-# the constructors will appear in the respective orders defined by
-# SORT_MEMBER_DOCS and SORT_BRIEF_DOCS.
-# This tag will be ignored for brief docs if SORT_BRIEF_DOCS is set to NO
-# and ignored for detailed docs if SORT_MEMBER_DOCS is set to NO.
-
-SORT_MEMBERS_CTORS_1ST = NO
-
 # If the SORT_GROUP_NAMES tag is set to YES then doxygen will sort the
 # hierarchy of group names into alphabetical order. If set to NO (the default)
 # the group names will appear in their defined order.
@@ -489,6 +446,12 @@ 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.
@@ -496,8 +459,7 @@ SHOW_USED_FILES        = YES
 SHOW_FILES             = YES

 # Set the SHOW_NAMESPACES tag to NO to disable the generation of the
-# Namespaces page.
-# This will remove the Namespaces entry from the Quick Index
+# Namespaces page.  This will remove the Namespaces entry from the Quick Index
 # and from the Folder Tree View (if specified). The default is YES.

 SHOW_NAMESPACES        = YES
@@ -512,15 +474,6 @@ SHOW_NAMESPACES        = YES

 FILE_VERSION_FILTER    =

-# The LAYOUT_FILE tag can be used to specify a layout file which will be parsed
-# by doxygen. The layout file controls the global structure of the generated
-# output files in an output format independent way. The create the layout file
-# that represents doxygen's defaults, run doxygen with the -l option.
-# You can optionally specify a file name after the option, if omitted
-# DoxygenLayout.xml will be used as the name of the layout file.
-
-LAYOUT_FILE            =
-
 #---------------------------------------------------------------------------
 # configuration options related to warning and progress messages
 #---------------------------------------------------------------------------
@@ -624,8 +577,7 @@ EXCLUDE_SYMLINKS       = NO
 # against the file with absolute path, so to exclude all test directories
 # for example use the pattern */test/*

-EXCLUDE_PATTERNS       = *.git \
-                         *.d
+EXCLUDE_PATTERNS       = *.git *.d

 # The EXCLUDE_SYMBOLS tag can be used to specify one or more symbol names
 # (namespaces, classes, functions, etc.) that should be excluded from the
@@ -639,14 +591,14 @@ EXCLUDE_SYMBOLS        =
 # directories that contain example code fragments that are included (see
 # the \include command).

-EXAMPLE_PATH           = doc/examples/
+EXAMPLE_PATH           =

 # 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       =

 # 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
@@ -666,17 +618,14 @@ IMAGE_PATH             =
 # by executing (via popen()) the command <filter> <input-file>, where <filter>
 # is the value of the INPUT_FILTER tag, and <input-file> is the name of an
 # input file. Doxygen will then use the output that the filter program writes
-# to standard output.
-# If FILTER_PATTERNS is specified, this tag will be
+# to standard output.  If FILTER_PATTERNS is specified, this tag will be
 # ignored.

 INPUT_FILTER           =

 # The FILTER_PATTERNS tag can be used to specify filters on a per file pattern
-# basis.
-# Doxygen will compare the file name with each pattern and apply the
-# filter if there is a match.
-# The filters are a list of the form:
+# basis.  Doxygen will compare the file name with each pattern and apply the
+# filter if there is a match.  The filters are a list of the form:
 # pattern=filter (like *.cpp=my_cpp_filter). See INPUT_FILTER for further
 # info on how filters are used. If FILTER_PATTERNS is empty, INPUT_FILTER
 # is applied to all files.
@@ -726,8 +675,7 @@ REFERENCES_RELATION    = NO
 # If the REFERENCES_LINK_SOURCE tag is set to YES (the default)
 # and SOURCE_BROWSER tag is set to YES, then the hyperlinks from
 # functions in REFERENCES_RELATION and REFERENCED_BY_RELATION lists will
-# link to the source code.
-# Otherwise they will link to the documentation.
+# link to the source code.  Otherwise they will link to the documentstion.

 REFERENCES_LINK_SOURCE = YES

@@ -759,7 +707,7 @@ ALPHABETICAL_INDEX     = YES
 # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
 # in which this list will be split (can be a number in the range [1..20])

-COLS_IN_ALPHA_INDEX    = 2
+COLS_IN_ALPHA_INDEX    = 5

 # In case all classes in a project start with a common prefix, all
 # classes will be put under the same header in the alphabetical index.
@@ -793,13 +741,13 @@ 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            =

 # 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            =

 # 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
@@ -808,57 +756,29 @@ 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        =

-# The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
-# Doxygen will adjust the colors in the stylesheet and background images
-# according to this color. Hue is specified as an angle on a colorwheel,
-# see http://en.wikipedia.org/wiki/Hue for more information.
-# For instance the value 0 represents red, 60 is yellow, 120 is green,
-# 180 is cyan, 240 is blue, 300 purple, and 360 is red again.
-# The allowed range is 0 to 359.
+# 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_COLORSTYLE_HUE    = 120
+HTML_ALIGN_MEMBERS     = YES

-# 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
-# grayscales only. A value of 255 will produce the most vivid colors.
+# If the GENERATE_HTMLHELP tag is set to YES, additional index files
+# will be generated that can be used as input for tools like the
+# Microsoft HTML help workshop to generate a compiled HTML help file (.chm)
+# of the generated HTML documentation.

-HTML_COLORSTYLE_SAT    = 100
-
-# The HTML_COLORSTYLE_GAMMA tag controls the gamma correction applied to
-# the luminance component of the colors in the HTML output. Values below
-# 100 gradually make the output lighter, whereas values above 100 make
-# the output darker. The value divided by 100 is the actual gamma applied,
-# so 80 represents a gamma of 0.8, The value 220 represents a gamma of 2.2,
-# and 100 does not change the gamma.
-
-HTML_COLORSTYLE_GAMMA  = 80
-
-# If the HTML_TIMESTAMP tag is set to YES then the footer of each generated HTML
-# page will contain the date and time when the page was generated. Setting
-# this to NO can help when comparing the output of multiple runs.
-
-HTML_TIMESTAMP         = 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
-# JavaScript and DHTML is required (for instance Mozilla 1.0+, Firefox
-# Netscape 6.0+, Internet explorer 5.0+, Konqueror, or Safari).
-
-HTML_DYNAMIC_SECTIONS  = NO
+GENERATE_HTMLHELP      = NO

 # If the GENERATE_DOCSET tag is set to YES, additional index files
 # will be generated that can be used as input for Apple's Xcode 3
-# integrated development environment, introduced with OS X 10.5 (Leopard).
+# integrated development environment, introduced with OSX 10.5 (Leopard).
 # To create a documentation set, doxygen will generate a Makefile in the
 # HTML output directory. Running make will produce the docset in that
 # directory and running "make install" will install the docset in
 # ~/Library/Developer/Shared/Documentation/DocSets so that Xcode will find
 # it at startup.
-# See http://developer.apple.com/tools/creatingdocsetswithdoxygen.html
-# for more information.

 GENERATE_DOCSET        = NO

@@ -876,22 +796,13 @@ DOCSET_FEEDNAME        = "Doxygen generated docs"

 DOCSET_BUNDLE_ID       = org.doxygen.Project

-# When GENERATE_PUBLISHER_ID tag specifies a string that should uniquely identify
-# the documentation publisher. This should be a reverse domain-name style
-# string, e.g. com.mycompany.MyDocSet.documentation.
+# 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
+# JavaScript and DHTML is required (for instance Mozilla 1.0+, Firefox
+# Netscape 6.0+, Internet explorer 5.0+, Konqueror, or Safari).

-DOCSET_PUBLISHER_ID    = org.doxygen.Publisher
-
-# The GENERATE_PUBLISHER_NAME tag identifies the documentation publisher.
-
-DOCSET_PUBLISHER_NAME  = Publisher
-
-# If the GENERATE_HTMLHELP tag is set to YES, additional index files
-# will be generated that can be used as input for tools like the
-# Microsoft HTML help workshop to generate a compiled HTML help file (.chm)
-# of the generated HTML documentation.
-
-GENERATE_HTMLHELP      = NO
+HTML_DYNAMIC_SECTIONS  = NO

 # If the GENERATE_HTMLHELP tag is set to YES, the CHM_FILE tag can
 # be used to specify the file name of the resulting .chm file. You
@@ -930,76 +841,6 @@ BINARY_TOC             = NO

 TOC_EXPAND             = NO

-# If the GENERATE_QHP tag is set to YES and both QHP_NAMESPACE and
-# QHP_VIRTUAL_FOLDER are set, an additional index file will be generated
-# that can be used as input for Qt's qhelpgenerator to generate a
-# Qt Compressed Help (.qch) of the generated HTML documentation.
-
-GENERATE_QHP           = NO
-
-# If the QHG_LOCATION tag is specified, the QCH_FILE tag can
-# be used to specify the file name of the resulting .qch file.
-# The path specified is relative to the HTML output folder.
-
-QCH_FILE               =
-
-# The QHP_NAMESPACE tag specifies the namespace to use when generating
-# Qt Help Project output. For more information please see
-# http://doc.trolltech.com/qthelpproject.html#namespace
-
-QHP_NAMESPACE          = org.doxygen.Project
-
-# The QHP_VIRTUAL_FOLDER tag specifies the namespace to use when generating
-# Qt Help Project output. For more information please see
-# http://doc.trolltech.com/qthelpproject.html#virtual-folders
-
-QHP_VIRTUAL_FOLDER     = doc
-
-# If QHP_CUST_FILTER_NAME is set, it specifies the name of a custom filter to
-# add. For more information please see
-# http://doc.trolltech.com/qthelpproject.html#custom-filters
-
-QHP_CUST_FILTER_NAME   =
-
-# The QHP_CUST_FILT_ATTRS tag specifies the list of the attributes of the
-# custom filter to add. For more information please see
-# <a href="http://doc.trolltech.com/qthelpproject.html#custom-filters">
-# Qt Help Project / Custom Filters</a>.
-
-QHP_CUST_FILTER_ATTRS  =
-
-# The QHP_SECT_FILTER_ATTRS tag specifies the list of the attributes this
-# project's
-# filter section matches.
-# <a href="http://doc.trolltech.com/qthelpproject.html#filter-attributes">
-# Qt Help Project / Filter Attributes</a>.
-
-QHP_SECT_FILTER_ATTRS  =
-
-# If the GENERATE_QHP tag is set to YES, the QHG_LOCATION tag can
-# be used to specify the location of Qt's qhelpgenerator.
-# If non-empty doxygen will try to run qhelpgenerator on the generated
-# .qhp file.
-
-QHG_LOCATION           =
-
-# If the GENERATE_ECLIPSEHELP tag is set to YES, additional index files
-#  will be generated, which together with the HTML files, form an Eclipse help
-# plugin. To install this plugin and make it available under the help contents
-# menu in Eclipse, the contents of the directory containing the HTML and XML
-# files needs to be copied into the plugins directory of eclipse. The name of
-# the directory within the plugins directory should be the same as
-# the ECLIPSE_DOC_ID value. After copying Eclipse needs to be restarted before
-# the help appears.
-
-GENERATE_ECLIPSEHELP   = NO
-
-# A unique identifier for the eclipse help plugin. When installing the plugin
-# the directory name containing the HTML and XML files should also have
-# this name.
-
-ECLIPSE_DOC_ID         = org.doxygen.Project
-
 # The DISABLE_INDEX tag can be used to turn on/off the condensed index at
 # top of each HTML page. The value NO (the default) enables the index and
 # the value YES disables it.
@@ -1013,11 +854,18 @@ ENUM_VALUES_PER_LINE   = 4

 # The GENERATE_TREEVIEW tag is used to specify whether a tree-like index
 # structure should be generated to display hierarchical information.
-# If the tag value is set to YES, a side panel will be generated
+# If the tag value is set to FRAME, a side panel will be generated
 # containing a tree-like index structure (just like the one that
 # is generated for HTML Help). For this to work a browser that supports
-# JavaScript, DHTML, CSS and frames is required (i.e. any modern browser).
-# Windows users are probably better off using the HTML help feature.
+# JavaScript, DHTML, CSS and frames is required (for instance Mozilla 1.0+,
+# Netscape 6.0+, Internet explorer 5.0+, or Konqueror). Windows users are
+# probably better off using the HTML help feature. Other possible values
+# for this tag are: HIERARCHIES, which will generate the Groups, Directories,
+# and Class Hiererachy pages using a tree view instead of an ordered list;
+# ALL, which combines the behavior of FRAME and HIERARCHIES; and NONE, which
+# disables this behavior completely. For backwards compatibility with previous
+# releases of Doxygen, the values YES and NO are equivalent to FRAME and NONE
+# respectively.

 GENERATE_TREEVIEW      = NO

@@ -1027,11 +875,6 @@ GENERATE_TREEVIEW      = NO

 TREEVIEW_WIDTH         = 250

-# When the EXT_LINKS_IN_WINDOW option is set to YES doxygen will open
-# links to external symbols imported via tag files in a separate window.
-
-EXT_LINKS_IN_WINDOW    = NO
-
 # Use this tag to change the font size of Latex formulas included
 # as images in the HTML documentation. The default is 10. Note that
 # when you change the font size after a successful doxygen run you need
@@ -1040,34 +883,6 @@ EXT_LINKS_IN_WINDOW    = NO

 FORMULA_FONTSIZE       = 10

-# Use the FORMULA_TRANPARENT tag to determine whether or not the images
-# generated for formulas are transparent PNGs. Transparent PNGs are
-# not supported properly for IE 6.0, but are supported on all modern browsers.
-# Note that when changing this option you need to delete any form_*.png files
-# in the HTML output before the changes have effect.
-
-FORMULA_TRANSPARENT    = YES
-
-# When the SEARCHENGINE tag is enabled doxygen will generate a search box
-# for the HTML output. The underlying search engine uses javascript
-# and DHTML and should work on any modern browser. Note that when using
-# HTML help (GENERATE_HTMLHELP), Qt help (GENERATE_QHP), or docsets
-# (GENERATE_DOCSET) there is already a search function so this one should
-# typically be disabled. For large projects the javascript based search engine
-# can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution.
-
-SEARCHENGINE           = NO
-
-# When the SERVER_BASED_SEARCH tag is enabled the search engine will be
-# implemented using a PHP enabled web server instead of at the web client
-# using Javascript. Doxygen will generate the search PHP script and index
-# file to put on the web server. The advantage of the server
-# based approach is that it scales better to large projects and allows
-# full text search. The disadvances is that it is more difficult to setup
-# and does not have live searching capabilities.
-
-SERVER_BASED_SEARCH    = NO
-
 #---------------------------------------------------------------------------
 # configuration options related to the LaTeX output
 #---------------------------------------------------------------------------
@@ -1085,9 +900,6 @@ LATEX_OUTPUT           = latex

 # The LATEX_CMD_NAME tag can be used to specify the LaTeX command name to be
 # invoked. If left blank `latex' will be used as the default command name.
-# Note that when enabling USE_PDFLATEX this option is only used for
-# generating bitmaps for formulas in the HTML output, but not in the
-# Makefile that is written to the output directory.

 LATEX_CMD_NAME         = latex

@@ -1147,13 +959,6 @@ LATEX_BATCHMODE        = NO

 LATEX_HIDE_INDICES     = NO

-# If LATEX_SOURCE_CODE is set to YES then doxygen will include
-# source code with syntax highlighting in the LaTeX output.
-# Note that which sources are shown also depends on other settings
-# such as SOURCE_BROWSER.
-
-LATEX_SOURCE_CODE      = NO
-
 #---------------------------------------------------------------------------
 # configuration options related to the RTF output
 #---------------------------------------------------------------------------
@@ -1290,10 +1095,8 @@ GENERATE_PERLMOD       = NO
 PERLMOD_LATEX          = NO

 # If the PERLMOD_PRETTY tag is set to YES the Perl module output will be
-# nicely formatted so it can be parsed by a human reader.
-# This is useful
-# if you want to understand what is going on.
-# On the other hand, if this
+# nicely formatted so it can be parsed by a human reader.  This is useful
+# if you want to understand what is going on.  On the other hand, if this
 # tag is set to NO the size of the Perl module output will be much smaller
 # and Perl will parse it just the same.

@@ -1355,18 +1158,16 @@ INCLUDE_FILE_PATTERNS  =
 # undefined via #undef or recursively expanded use the := operator
 # instead of the = operator.

-PREDEFINED             = "__attribute__(x)=" \
-                         "DECLARE_ALIGNED(a,t,n)=t n" \
-                         "offsetof(x,y)=0x42" \
-                         av_alloc_size \
+PREDEFINED             = __attribute__(x)="" "RENAME(x)=x ## _TMPL" "DEF(x)=x ## _TMPL" \
+                         HAVE_AV_CONFIG_H HAVE_MMX HAVE_MMX2 HAVE_AMD3DNOW \

 # 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.
 # The macro definition that is found in the sources will be used.
 # Use the PREDEFINED tag if you want to use a different macro definition.

-EXPAND_AS_DEFINED      = declare_idct  \
-                         READ_PAR_DATA \
+#EXPAND_AS_DEFINED      = FF_COMMON_FRAME
+EXPAND_AS_DEFINED      = declare_idct(idct, table, idct_row_head, idct_row, idct_row_tail, idct_row_mid)

 # If the SKIP_FUNCTION_MACROS tag is set to YES (the default) then
 # doxygen's preprocessor will remove all function-like macros that are alone
@@ -1384,11 +1185,9 @@ SKIP_FUNCTION_MACROS   = YES
 # Optionally an initial location of the external documentation
 # can be added for each tagfile. The format of a tag file without
 # this location is as follows:
-#
-# TAGFILES = file1 file2 ...
+#   TAGFILES = file1 file2 ...
 # Adding location for the tag files is done as follows:
-#
-# TAGFILES = file1=loc1 "file2 = loc2" ...
+#   TAGFILES = file1=loc1 "file2 = loc2" ...
 # where "loc1" and "loc2" can be relative or absolute paths or
 # URLs. If a location is present for each tag, the installdox tool
 # does not have to be run to correct the links.
@@ -1456,14 +1255,6 @@ HIDE_UNDOC_RELATIONS   = YES

 HAVE_DOT               = NO

-# The DOT_NUM_THREADS specifies the number of dot invocations doxygen is
-# allowed to run in parallel. When set to 0 (the default) doxygen will
-# base this on the number of processors available in the system. You can set it
-# explicitly to a value larger than 0 to get control over the balance
-# between CPU load and processing speed.
-
-DOT_NUM_THREADS        = 0
-
 # By default doxygen will write a font called FreeSans.ttf to the output
 # directory and reference it in all dot files that doxygen generates. This
 # font does not include all possible unicode characters however, so when you need
@@ -1475,11 +1266,6 @@ DOT_NUM_THREADS        = 0

 DOT_FONTNAME           = FreeSans

-# The DOT_FONTSIZE tag can be used to set the size of the font of dot graphs.
-# The default size is 10pt.
-
-DOT_FONTSIZE           = 10
-
 # By default doxygen will tell dot to use the output directory to look for the
 # FreeSans.ttf font (which doxygen will put there itself). If you specify a
 # different font using DOT_FONTNAME you can set the path where dot
@@ -1597,10 +1383,10 @@ DOT_GRAPH_MAX_NODES    = 50
 MAX_DOT_GRAPH_DEPTH    = 0

 # Set the DOT_TRANSPARENT tag to YES to generate images with a transparent
-# background. This is disabled by default, because dot on Windows does not
-# seem to support this out of the box. Warning: Depending on the platform used,
-# enabling this option may lead to badly anti-aliased labels on the edges of
-# a graph (i.e. they become hard to read).
+# background. This is enabled by default, which results in a transparent
+# background. Warning: Depending on the platform used, enabling this option
+# may lead to badly anti-aliased labels on the edges of a graph (i.e. they
+# become hard to read).

 DOT_TRANSPARENT        = YES

@@ -1622,3 +1408,12 @@ GENERATE_LEGEND        = YES
 # the various graphs.

 DOT_CLEANUP            = YES
+
+#---------------------------------------------------------------------------
+# Configuration::additions related to the search engine
+#---------------------------------------------------------------------------
+
+# The SEARCHENGINE tag specifies whether or not a search engine should be
+# used. If set to NO the values of all tags below this one will be ignored.
+
+SEARCHENGINE           = NO
--- a/73
+++ b/73
@@ -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 library libfaac 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.
--- a/119
+++ b/119
@@ -4,12 +4,6 @@ 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!
-
-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
-and related discussions.
-

 Project Leader
 ==============
@@ -25,7 +19,7 @@ ffmpeg:
  ffmpeg.c                              Michael Niedermayer

 ffplay:
-  ffplay.c                              Marton Balint
+  ffplay.c                              Michael Niedermayer

 ffprobe:
  ffprobe.c                             Stefano Sabatini
@@ -44,13 +38,13 @@ Miscellaneous Areas
 ===================

 documentation                           Mike Melanson
-website                                 Robert Swain, Lou Logan
+website                                 Robert Swain
 build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
-mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
+project server                          Diego Biurrun, Mans Rullgard
+mailinglists                            Michael Niedermayer, Baptiste Coudurier
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
-release management                      Michael Niedermayer
+release management                      Diego Biurrun, Reinhard Tartler


 libavutil
@@ -87,8 +81,6 @@ Generic Parts:
    bitstream.c, bitstream.h            Michael Niedermayer
  CABAC:
    cabac.h, cabac.c                    Michael Niedermayer
-  codec names:
-    codec_names.sh                      Nicolas George
  DSP utilities:
    dsputils.c, dsputils.h              Michael Niedermayer
  entropy coding:
@@ -130,12 +122,9 @@ 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
@@ -145,7 +134,6 @@ Codecs:
  dv.c                                  Roman Shaposhnik
  eacmv*, eaidct*, eat*                 Peter Ross
  ffv1.c                                Michael Niedermayer
-  ffwavesynth.c                         Nicolas George
  flac*                                 Justin Ruggles
  flashsv*                              Benjamin Larsson
  flicvideo.c                           Mike Melanson
@@ -162,20 +150,16 @@ 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
  lcl*.c                                Roberto Togni, Reimar Doeffinger
-  libcelt_dec.c                         Nicolas George
  libgsm.c                              Michel Bardiaux
  libdirac*                             David Conrad
  libopenjpeg.c                         Jaikrishnan Menon
-  libopenjpegenc.c                      Michael Bradshaw
  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 +167,7 @@ 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 +178,6 @@ 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 +202,6 @@ 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
@@ -229,13 +211,9 @@ Codecs:
  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
  txd.c                                 Ivo van Poorten
  ulti*                                 Kostya Shishkov
-  v410*.c                               Derek Buitenhuis
  vb.c                                  Kostya Shishkov
-  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
@@ -250,19 +228,14 @@ 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:
  crystalhd.c                           Philip Langdale
  dxva2*                                Laurent Aimar
-  libstagefright.cpp                    Mohamed Naufal
  vaapi*                                Gwenole Beauchesne
-  vda*                                  Sebastien Zwickert
  vdpau*                                Carl Eugen Hoyos


@@ -272,27 +245,10 @@ 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:
-  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
-  vf_yadif.c                            Michael Niedermayer
-
-Sources:
-  vsrc_mandelbrot.c                     Michael Niedermayer

 libavformat
 ===========
@@ -307,25 +263,17 @@ 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 +284,17 @@ 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
@@ -362,8 +305,6 @@ Muxers/Demuxers:
  msnwc_tcp.c                           Ramiro Polla
  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 +312,8 @@ 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
@@ -385,15 +324,12 @@ Muxers/Demuxers:
  rtpdec_asf.*                          Ronald S. Bultje
  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
  rtsp.c                                Luca Barbato
-  sbgdec.c                              Nicolas George
  sdp.c                                 Martin Storsjo
  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 +338,13 @@ 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
 =====================================

@@ -442,18 +362,8 @@ Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer


-Releases
-========
-
-1.2                                     Michael Niedermayer
-1.1                                     Michael Niedermayer
-1.0                                     Michael Niedermayer
-
-If you want to maintain an older release, please contact us
-
-
-GnuPG Fingerprints of maintainers and contributors
-==================================================
+GnuPG Fingerprints of maintainers and others who have svn write access
+======================================================================

 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
@@ -461,24 +371,19 @@ 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
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
 Justin Ruggles                3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
-Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
-Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
+Panagiotis Issaris            515C E262 10A8 FDCE 5481 7B9C 3AD7 D9A5 071D B3A9
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
 Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 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
--- a/282
+++ b/282
@@ -1,80 +1,72 @@
-MAIN_MAKEFILE=1
 include config.mak

-vpath %.c    $(SRC_PATH)
-vpath %.cpp  $(SRC_PATH)
-vpath %.h    $(SRC_PATH)
-vpath %.S    $(SRC_PATH)
-vpath %.asm  $(SRC_PATH)
-vpath %.v    $(SRC_PATH)
-vpath %.texi $(SRC_PATH)
-vpath %/fate_config.sh.template $(SRC_PATH)
+SRC_DIR = $(SRC_PATH_BARE)
+
+vpath %.c   $(SRC_DIR)
+vpath %.h   $(SRC_DIR)
+vpath %.S   $(SRC_DIR)
+vpath %.asm $(SRC_DIR)
+vpath %.v   $(SRC_DIR)
+vpath %.texi $(SRC_PATH_BARE)

 PROGS-$(CONFIG_FFMPEG)   += ffmpeg
 PROGS-$(CONFIG_FFPLAY)   += ffplay
 PROGS-$(CONFIG_FFPROBE)  += ffprobe
 PROGS-$(CONFIG_FFSERVER) += ffserver

-PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-
-OBJS        = cmdutils.o $(EXEOBJS)
-OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
+PROGS      := $(PROGS-yes:%=%$(EXESUF))
+PROGS_G     = $(PROGS-yes:%=%_g$(EXESUF))
+OBJS        = $(PROGS-yes:%=%.o) cmdutils.o
+MANPAGES    = $(PROGS-yes:%=doc/%.1)
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html)
+TOOLS       = $(addprefix tools/, $(addsuffix $(EXESUF), cws2fws graph2dot lavfi-showfiltfmts pktdumper probetest qt-faststart trasher))
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher
-TOOLS-$(CONFIG_ZLIB) += cws2fws
+HOSTPROGS  := $(TESTTOOLS:%=tests/%)

 BASENAMES   = ffmpeg ffplay ffprobe ffserver
-ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+ALLPROGS    = $(BASENAMES:%=%$(EXESUF))
+ALLPROGS_G  = $(BASENAMES:%=%_g$(EXESUF))
 ALLMANPAGES = $(BASENAMES:%=%.1)

+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
+
 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-$(CONFIG_SWRESAMPLE)+= swresample
 FFLIBS-$(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
+DATA_FILES := $(wildcard $(SRC_DIR)/ffpresets/*.ffpreset)

 SKIPHEADERS = cmdutils_common_opts.h

-include $(SRC_PATH)/common.mak
+include common.mak

+FF_LDFLAGS   := $(FFLDFLAGS)
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)

-all: $(PROGS)
+all-$(CONFIG_DOC): documentation
+
+all: $(FF_DEP_LIBS) $(PROGS)

 $(PROGS): %$(EXESUF): %_g$(EXESUF)
 	$(CP) $< $@
 	$(STRIP) $@

-$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
-
-tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
-
 config.h: .config
-.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
+.config: $(wildcard $(FFLIBS:%=$(SRC_DIR)/lib%/all*.c))
 	@-tput bold 2>/dev/null
 	@-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

 define RESET
 $(1) :=
@@ -84,33 +76,31 @@ endef
 define DOSUBDIR
 $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
-include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/library.mak
+include $(1)/Makefile
 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): FF_LDFLAGS += $(FFSERVERLDFLAGS)

-$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
+%_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
+	$(LD) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)

-%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+alltools: $(TOOLS)

-OBJDIRS += tools
+tools/%$(EXESUF): tools/%.o
+	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
+
+tools/%.o: tools/%.c
+	$(CC) $(CPPFLAGS) $(CFLAGS) -c $(CC_O) $<

 -include $(wildcard tools/*.d)
+-include $(wildcard tests/*.d)

-VERSION_SH  = $(SRC_PATH)/version.sh
-GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
+VERSION_SH  = $(SRC_PATH_BARE)/version.sh
+GIT_LOG     = $(SRC_PATH_BARE)/.git/logs/HEAD

 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
@@ -122,6 +112,28 @@ version.h .version:
 # force version.sh to run whenever version might have changed
 -include .version

+DOCS = $(addprefix doc/, developer.html faq.html general.html libavfilter.html) $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
+
+documentation: $(DOCS)
+
+-include $(wildcard $(DOCS:%=%.d))
+
+TEXIDEP = awk '/^@include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
+
+doc/%.html: TAG = HTML
+doc/%.html: doc/%.texi $(SRC_PATH_BARE)/doc/t2h.init
+	$(Q)$(TEXIDEP)
+	$(M)texi2html -monolithic --init-file $(SRC_PATH_BARE)/doc/t2h.init --output $@ $<
+
+doc/%.pod: TAG = POD
+doc/%.pod: doc/%.texi
+	$(Q)$(TEXIDEP)
+	$(M)doc/texi2pod.pl $< $@
+
+doc/%.1: TAG = MAN
+doc/%.1: doc/%.pod
+	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
+
 ifdef PROGS
 install: install-progs install-data
 endif
@@ -131,18 +143,22 @@ install: install-libs install-headers
 install-libs: install-libs-yes

 install-progs-yes:
+install-progs-$(CONFIG_DOC): install-man
 install-progs-$(CONFIG_SHARED): install-libs

 install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
-	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
+	$(INSTALL) -c -m 755 $(PROGS) "$(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
+install-man: $(MANPAGES)
+	$(Q)mkdir -p "$(MANDIR)/man1"
+	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
+
+uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data uninstall-man

 uninstall-progs:
 	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
@@ -150,44 +166,142 @@ uninstall-progs:
 uninstall-data:
 	$(RM) -r "$(DATADIR)"

-clean::
+uninstall-man:
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
+
+testclean:
+	$(RM) -r tests/vsynth1 tests/vsynth2 tests/data
+	$(RM) $(addprefix tests/,$(CLEANSUFFIXES))
+	$(RM) tests/seek_test$(EXESUF) tests/seek_test.o
+	$(RM) $(TESTTOOLS:%=tests/%$(HOSTEXESUF))
+
+clean:: testclean
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
+	$(RM) doc/*.html doc/*.pod doc/*.1
+	$(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)

-# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
-coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
-	$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
+# regression tests

-coverage-html: coverage.info
-	$(Q)mkdir -p $@
-	$(Q)genhtml -o $@ $<
-	$(Q)touch $@
+check: test

-check: all alltools examples testprogs fate
+fulltest test: codectest lavftest lavfitest seektest

-include $(SRC_PATH)/doc/Makefile
-include $(SRC_PATH)/tests/Makefile
+FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref

-$(sort $(OBJDIRS)):
-	$(Q)mkdir -p $@
+codectest: fate-codec
+lavftest:  fate-lavf
+lavfitest: fate-lavfi
+seektest:  fate-seek

-# Dummy rule to stop make trying to rebuild removed or renamed headers
-%.h:
-	@:
+AREF = fate-acodec-aref
+VREF = fate-vsynth1-vref fate-vsynth2-vref
+REFS = $(AREF) $(VREF)

-# Disable suffix rules.  Most of the builtin rules are suffix rules,
-# so this saves some time on slow systems.
-.SUFFIXES:
+$(VREF): ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm
+$(AREF): ffmpeg$(EXESUF) tests/data/asynth1.sw

-.PHONY: all all-yes alltools check *clean config install*
-.PHONY: testprogs uninstall*
+ffservertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/data/asynth1.sw
+	@echo
+	@echo "Unfortunately ffserver is broken and therefore its regression"
+	@echo "test fails randomly. Treat the results accordingly."
+	@echo
+	$(SRC_PATH)/tests/ffserver-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/ffserver.conf
+
+tests/vsynth1/00.pgm: tests/videogen$(HOSTEXESUF)
+	@mkdir -p tests/vsynth1
+	$(M)./$< 'tests/vsynth1/'
+
+tests/vsynth2/00.pgm: tests/rotozoom$(HOSTEXESUF)
+	@mkdir -p tests/vsynth2
+	$(M)./$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
+
+tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF)
+	@mkdir -p tests/data
+	$(M)./$< $@
+
+tests/data/asynth1.sw tests/vsynth%/00.pgm: TAG = GEN
+
+tests/seek_test$(EXESUF): tests/seek_test.o $(FF_DEP_LIBS)
+	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
+
+tools/lavfi-showfiltfmts$(EXESUF): tools/lavfi-showfiltfmts.o $(FF_DEP_LIBS)
+	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
+
+include $(SRC_PATH_BARE)/tests/fate.mak
+include $(SRC_PATH_BARE)/tests/fate2.mak
+
+include $(SRC_PATH_BARE)/tests/fate/aac.mak
+include $(SRC_PATH_BARE)/tests/fate/als.mak
+include $(SRC_PATH_BARE)/tests/fate/fft.mak
+include $(SRC_PATH_BARE)/tests/fate/h264.mak
+include $(SRC_PATH_BARE)/tests/fate/mp3.mak
+include $(SRC_PATH_BARE)/tests/fate/vorbis.mak
+include $(SRC_PATH_BARE)/tests/fate/vp8.mak
+
+FATE_ACODEC  = $(ACODEC_TESTS:%=fate-acodec-%)
+FATE_VSYNTH1 = $(VCODEC_TESTS:%=fate-vsynth1-%)
+FATE_VSYNTH2 = $(VCODEC_TESTS:%=fate-vsynth2-%)
+FATE_VCODEC  = $(FATE_VSYNTH1) $(FATE_VSYNTH2)
+FATE_LAVF    = $(LAVF_TESTS:%=fate-lavf-%)
+FATE_LAVFI   = $(LAVFI_TESTS:%=fate-lavfi-%)
+FATE_SEEK    = $(SEEK_TESTS:seek_%=fate-seek-%)
+
+FATE = $(FATE_ACODEC)                                                   \
+       $(FATE_VCODEC)                                                   \
+       $(FATE_LAVF)                                                     \
+       $(FATE_LAVFI)                                                    \
+       $(FATE_SEEK)                                                     \
+
+$(filter-out %-aref,$(FATE_ACODEC)): $(AREF)
+$(filter-out %-vref,$(FATE_VCODEC)): $(VREF)
+$(FATE_LAVF):   $(REFS)
+$(FATE_LAVFI):  $(REFS) tools/lavfi-showfiltfmts$(EXESUF)
+$(FATE_SEEK):   fate-codec fate-lavf tests/seek_test$(EXESUF)
+
+$(FATE_ACODEC):  CMD = codectest acodec
+$(FATE_VSYNTH1): CMD = codectest vsynth1
+$(FATE_VSYNTH2): CMD = codectest vsynth2
+$(FATE_LAVF):    CMD = lavftest
+$(FATE_LAVFI):   CMD = lavfitest
+$(FATE_SEEK):    CMD = seektest
+
+fate-codec:  fate-acodec fate-vcodec
+fate-acodec: $(FATE_ACODEC)
+fate-vcodec: $(FATE_VCODEC)
+fate-lavf:   $(FATE_LAVF)
+fate-lavfi:  $(FATE_LAVFI)
+fate-seek:   $(FATE_SEEK)
+
+ifdef SAMPLES
+FATE += $(FATE_TESTS)
+fate-rsync:
+	rsync -vaLW rsync://fate-suite.libav.org/fate-suite/ $(SAMPLES)
+else
+fate-rsync:
+	@echo "use 'make fate-rsync SAMPLES=/path/to/samples' to sync the fate suite"
+$(FATE_TESTS):
+	@echo "SAMPLES not specified, cannot run FATE. See doc/fate.txt for more information."
+endif
+
+FATE_UTILS = base64 tiny_psnr
+
+fate: $(FATE)
+
+$(FATE): ffmpeg$(EXESUF) $(FATE_UTILS:%=tests/%$(HOSTEXESUF))
+	@echo "TEST    $(@:fate-%=%)"
+	$(Q)$(SRC_PATH)/tests/fate-run.sh $@ "$(SAMPLES)" "$(TARGET_EXEC)" "$(TARGET_PATH)" '$(CMD)' '$(CMP)' '$(REF)' '$(FUZZ)' '$(THREADS)' '$(THREAD_TYPE)'
+
+fate-list:
+	@printf '%s\n' $(sort $(FATE))
+
+.PHONY: all alltools *clean check config documentation examples install*
+.PHONY: *test testprogs uninstall*
--- a/8
+++ b/8
@@ -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.
--- a/2
+++ b/2
@@ -1 +1 @@
-1.1.14
+0.7
--- a/1
+++ b/1
@@ -0,0 +1 @@
+0.7.1
--- a/arch.mak
+++ b/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)
--- a/cmdutils.c
+++ b/cmdutils.c
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -43,16 +43,11 @@ extern const char program_name[];
 */
 extern const int program_birth_year;

-/**
- * this year, defined by the program for show_banner()
- */
-extern const int this_year;
-
+extern const char **opt_names;
 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;
+extern AVDictionary *format_opts, *video_opts, *audio_opts, *sub_opts;

 /**
 * Initialize the cmdutils option system, in particular
@@ -67,7 +62,7 @@ void uninit_opts(void);

 /**
 * Trivial log callback.
- * Only suitable for opt_help and similar since it lacks prefix handling.
+ * Only suitable for show_help and similar since it lacks prefix handling.
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);

@@ -75,25 +70,17 @@ 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_report(const char *opt);
-
-int opt_max_alloc(void *optctx, 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_loglevel(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.
@@ -101,15 +88,14 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg);
 * parsed or the corresponding value is invalid.
 *
 * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
 * @param numstr the string to be parsed
 * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
 * string should be parsed
 * @param min the minimum valid accepted value
 * @param max the maximum valid accepted value
 */
-double parse_number_or_die(const char *context, const char *numstr, int type,
-                           double min, double max);
+double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max);

 /**
 * Parse a string specifying a time and return its corresponding
@@ -117,7 +103,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
 * the string cannot be correctly parsed.
 *
 * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
 * @param timestr the string to be parsed
 * @param is_duration a flag which tells how to interpret timestr, if
 * not zero timestr is interpreted as a duration, otherwise as a
@@ -125,21 +111,9 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
 *
 * @see parse_date()
 */
-int64_t parse_time_or_die(const char *context, const char *timestr,
-                          int is_duration);
+int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);

-typedef struct SpecifierOpt {
-    char *specifier;    /**< stream/chapter/program/... specifier */
-    union {
-        uint8_t *str;
-        int        i;
-        int64_t  i64;
-        float      f;
-        double   dbl;
-    } u;
-} SpecifierOpt;
-
-typedef struct OptionDef {
+typedef struct {
    const char *name;
    int flags;
 #define HAS_ARG    0x0001
@@ -148,222 +122,38 @@ 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_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
-                                   an int containing element count in the array. */
-#define OPT_TIME  0x10000
-#define OPT_DOUBLE 0x20000
     union {
-        void *dst_ptr;
-        int (*func_arg)(void *, const char *, const char *);
-        size_t off;
+        int *int_arg;
+        char **str_arg;
+        float *float_arg;
+        int (*func_arg)(const char *, const char *);
+        int64_t *int64_arg;
    } 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);
-
-/**
- * Show help for all options with given flags in class and all its
- * children.
- */
-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);
+void show_help_options(const OptionDef *options, const char *msg, int mask, int value);

 /**
 * 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
 * argument without a leading option name flag. NULL if such arguments do
 * not have to be processed.
 */
-void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
-                   void (* parse_arg_function)(void *optctx, const char*));
+void parse_options(int argc, char **argv, const OptionDef *options,
+                   int (* parse_arg_function)(const char *opt, const char *arg));

-/**
- * Parse one given option.
- *
- * @return on success 1 if arg was consumed, 0 otherwise; negative number on error
- */
-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.
- *
- * @param s  Corresponding format context.
- * @param st Stream from s to be checked.
- * @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
- *
- * @return 1 if the stream matches, 0 if it doesn't, <0 on error
- */
-int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
-
-/**
- * Filter out options for given codec.
- *
- * 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);
-
-/**
- * Setup AVCodecContext options for avformat_find_stream_info().
- *
- * Create an array of dictionaries, one dictionary for each stream
- * contained in s.
- * Each dictionary will contain the options from codec_opts which can
- * be applied to the corresponding stream codec context.
- *
- * @return pointer to the created array of dictionaries, NULL if it
- * cannot be created
- */
-AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
-                                           AVDictionary *codec_opts);
+void set_context_opts(void *ctx, void *opts_ctx, int flags, AVCodec *codec);

 /**
 * Print an error message to stderr, indicating filename and a human
@@ -381,89 +171,56 @@ void print_error(const char *filename, int err);
 * current version of the repository and of the libav* libraries used by
 * the program.
 */
-void show_banner(int argc, char **argv, const OptionDef *options);
+void show_banner(void);

 /**
 * Print the version of the program to stdout. The version message
 * depends on the current versions of the repository and of the libav*
 * libraries.
- * This option processing function does not utilize the arguments.
 */
-int show_version(void *optctx, const char *opt, const char *arg);
+void show_version(void);

 /**
 * 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);
+void show_license(void);

 /**
 * 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);
+void show_formats(void);

 /**
 * 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);
+void show_codecs(void);

 /**
 * 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);
+void show_filters(void);

 /**
 * 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);
+void show_bsfs(void);

 /**
 * 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);
+void show_protocols(void);

 /**
 * 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);
-
-/**
- * Print a listing containing all the sample formats supported by the
- * program.
- */
-int show_sample_fmts(void *optctx, const char *opt, const char *arg);
+void show_pix_fmts(void);

 /**
 * Return a positive value if a line read from standard input
@@ -475,13 +232,12 @@ 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
 * AVERROR error code in case of failure.
 */
-int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
+int read_file(const char *filename, char **bufptr, size_t *size);

 /**
 * Get a file corresponding to a preset file.
@@ -492,7 +248,7 @@ 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 +260,4 @@ 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);

-/**
- * Realloc array to hold new_size elements of elem_size.
- * Calls exit() 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 */
+#endif /* FFMPEG_CMDUTILS_H */
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -1,21 +1,13 @@
-    { "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" },
-    { "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" },
+    { "L", OPT_EXIT, {(void*)show_license}, "show license" },
+    { "h", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "?", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "help", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "-help", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "version", OPT_EXIT, {(void*)show_version}, "show version" },
+    { "formats"  , OPT_EXIT, {(void*)show_formats  }, "show available formats" },
+    { "codecs"   , OPT_EXIT, {(void*)show_codecs   }, "show available codecs" },
+    { "bsfs"     , OPT_EXIT, {(void*)show_bsfs     }, "show available bit stream filters" },
+    { "protocols", OPT_EXIT, {(void*)show_protocols}, "show available protocols" },
+    { "filters",   OPT_EXIT, {(void*)show_filters  }, "show available filters" },
+    { "pix_fmts" , OPT_EXIT, {(void*)show_pix_fmts }, "show available pixel formats" },
+    { "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
--- a/common.mak
+++ b/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 AS YASM AR LD HOSTCC STRIP CP
+SILENT = DEPCC YASMDEP RM RANLIB
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
@@ -21,46 +20,23 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif

-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+IFLAGS   := -I. -I$(SRC_PATH)
+CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
+CFLAGS   += $(ECFLAGS)
+YASMFLAGS += $(IFLAGS) -Pconfig.asm

-# 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)
-
-define COMPILE
-       $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
-endef
-
-COMPILE_C = $(call COMPILE,CC)
-COMPILE_CXX = $(call COMPILE,CXX)
-COMPILE_S = $(call COMPILE,AS)
+HOSTCFLAGS += $(IFLAGS)

 %.o: %.c
-	$(COMPILE_C)
-
-%.o: %.cpp
-	$(COMPILE_CXX)
-
-%.s: %.c
-	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
+	$(CCDEP)
+	$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_DEPFLAGS) -c $(CC_O) $<

 %.o: %.S
-	$(COMPILE_S)
+	$(ASDEP)
+	$(AS) $(CPPFLAGS) $(ASFLAGS) $(AS_DEPFLAGS) -c -o $@ $<

-%.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)/' $^ > $@
@@ -79,62 +55,37 @@ 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 := $(addprefix -l,$(addsuffix $(BUILDSUF),$(FFLIBS))) $(EXTRALIBS)
+FFLDFLAGS   := $(addprefix -Llib,$(ALLFFLIBS)) $(LDFLAGS)

-EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
-OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
-TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
-TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
-HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
-HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
-TOOLS     += $(TOOLS-yes)
-TOOLOBJS  := $(TOOLS:%=tools/%.o)
-TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
-HEADERS   += $(HEADERS-yes)
+EXAMPLES  := $(addprefix $(SUBDIR),$(addsuffix -example$(EXESUF),$(EXAMPLES)))
+OBJS      := $(addprefix $(SUBDIR),$(sort $(OBJS)))
+TESTOBJS  := $(addprefix $(SUBDIR),$(TESTOBJS) $(TESTPROGS:%=%-test.o))
+TESTPROGS := $(addprefix $(SUBDIR),$(addsuffix -test$(EXESUF),$(TESTPROGS)))
+HOSTOBJS  := $(addprefix $(SUBDIR),$(addsuffix .o,$(HOSTPROGS)))
+HOSTPROGS := $(addprefix $(SUBDIR),$(addsuffix $(HOSTEXESUF),$(HOSTPROGS)))

 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)
-
-alltools: $(TOOLS)
+SKIPHEADERS += $(addprefix $(ARCH)/,$(ARCH_HEADERS))
+SKIPHEADERS := $(addprefix $(SUBDIR),$(SKIPHEADERS-) $(SKIPHEADERS))
+checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))

 $(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))
-
-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
+CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver
 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))
--- a/compat/getopt.c
+++ b/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;
-}
--- a/compat/msvcrt/snprintf.c
+++ b/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;
-}
--- a/compat/msvcrt/snprintf.h
+++ b/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 */
--- a/compat/plan9/head
+++ b/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 ${n}q "$@"
--- a/compat/plan9/main.c
+++ b/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);
-}
--- a/compat/plan9/printf
+++ b/compat/plan9/printf
@@ -1,2 +0,0 @@
-#!/bin/sh
-exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"
--- a/compat/strtod.c
+++ b/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;
-}
--- a/compat/tms470/math.h
+++ b/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 }))
--- a/3314
+++ b/3314
--- a/doc/APIchanges
+++ b/doc/APIchanges
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -1,103 +0,0 @@
-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) $(LIBRARIES-yes:%=doc/%.html)     \
-              doc/developer.html                                        \
-              doc/faq.html                                              \
-              doc/fate.html                                             \
-              doc/general.html                                          \
-              doc/git-howto.html                                        \
-              doc/nut.html                                              \
-              doc/platform.html                                         \
-
-TXTPAGES    = doc/fate.txt                                              \
-
-
-DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
-DOCS-$(CONFIG_PODPAGES)  += $(PODPAGES)
-DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
-DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
-DOCS = $(DOCS-yes)
-
-all-$(CONFIG_DOC): doc
-
-doc: documentation
-
-apidoc: doc/doxy/html
-documentation: $(DOCS)
-
-TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
-
-doc/%.txt: TAG = TXT
-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)
-	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
-
-doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
-
-doc/%.1 doc/%.3: TAG = MAN
-doc/%.1: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
-doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
-
-$(DOCS) doc/doxy/html: | 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
-
-include $(wildcard $(DOCS:%=%.d))
-
-.PHONY: apidoc doc documentation
--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -1,13 +1,28 @@
 Release Notes
 =============

-* 1.1 "Fire Flower"  January, 2013
+* 0.8 "Love"  June, 2011
+* 0.7.1 "Peace" June, 2011 (identical to 0.8 but using 0.6 ABI/API)


 General notes
 -------------
-See the Changelog file for a list of significant changes. Note, there
-are many more new features and bugfixes than whats listed there.
+
+This release enables frame-based multithreaded decoding for a number of codecs,
+including theora, huffyuv, VP8, H.263, mpeg4 and H.264. Additionally, there has
+been a major cleanup of
+both internal and external APIs. For this reason, the major versions of all
+libraries except libpostproc have been bumped. This means that 0.8 can be installed
+side-by-side with previous releases, on the other hand applications need to be
+recompiled to use 0.8.
+
+Other important changes are more than 200 bugfixes, known regressions were fixed
+w.r.t 0.5 and 0.6, additions of decoders including, but not limited to,
+AMR-WB, single stream LATM/LOAS, G.722 ADPCM, a native VP8 decoder
+and HE-AACv2. Additionally, many new de/muxers such as WebM in Matroska, Apple
+HTTP Live Streaming, SAP, IEC 61937 (S/PDIF) have been added.
+
+See the Changelog file for a list of significant changes.

 Bugreports against FFmpeg git master or the most recent FFmpeg release are
 accepted. If you are experiencing issues with any formally released version of
@@ -15,11 +30,36 @@ 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.
+Note, if you have difficulty building for mingw, try --disable-outdev=sdl

-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
+-----------
+
+Please see git log of the public headers or the file doc/APIchanges for
+programmer-centric information. Note that some long-time deprecated APIs have
+been removed. Also, a number of additional APIs have been deprecated and might
+be removed in the next release.
+
+
+Other notable changes
+---------------------
+
+- high quality dithering in swscale to fix banding issues
+- ffmpeg is now interactive and various information can be turned on/off while its running
+- resolution changing support in ffmpeg
+- sdl output device
+- optimizations in libavfilter that make it much faster
+- split, buffer, select, lut, negate filters amongth others
+- more than 50 new video filters from mplayers libmpcodecs
+- many ARM NEON optimizations
+- nonfree libfaad support for AAC decoding removed
+- 4:4:4 H.264 decoding
+- 9/10bit H.264 decoding
+- Win64 Assembler support
+- native MMSH/MMST support
+- Windows TV demuxing
+- native AMR-WB decoding
+- native GSM-MS decoding
+- SMPTE 302M decoding
+- AVS encoding
--- a/doc/TODO
+++ b/doc/TODO
@@ -0,0 +1,82 @@
+ffmpeg TODO list:
+----------------
+
+Fabrice's TODO list: (unordered)
+-------------------
+Short term:
+
+- use AVFMTCTX_DISCARD_PKT in ffplay so that DV has a chance to work
+- add RTSP regression test (both client and server)
+- make ffserver allocate AVFormatContext
+- clean up (incompatible change, for 0.5.0):
+    * AVStream -> AVComponent
+    * AVFormatContext -> AVInputStream/AVOutputStream
+    * suppress rate_emu from AVCodecContext
+- add new float/integer audio filterting and conversion : suppress
+  CODEC_ID_PCM_xxc and use CODEC_ID_RAWAUDIO.
+- fix telecine and frame rate conversion
+
+Long term (ask me if you want to help):
+
+- commit new imgconvert API and new PIX_FMT_xxx alpha formats
+- commit new LGPL'ed float and integer-only AC3 decoder
+- add WMA integer-only decoder
+- add new MPEG4-AAC audio decoder (both integer-only and float version)
+
+Michael's TODO list: (unordered) (if anyone wanna help with sth, just ask)
+-------------------
+- optimize H264 CABAC
+- more optimizations
+- simper rate control
+
+Philip'a TODO list: (alphabetically ordered) (please help)
+------------------
+- Add a multi-ffm filetype so that feeds can be recorded into multiple files rather
+  than one big file.
+- Authenticated users support -- where the authentication is in the URL
+- Change ASF files so that the embedded timestamp in the frames is right rather
+  than being an offset from the start of the stream
+- Make ffm files more resilient to changes in the codec structures so that you
+  can play old ffm files.
+
+Baptiste's TODO list:
+-----------------
+- mov edit list support (AVEditList)
+- YUV 10 bit per component support "2vuy"
+- mxf muxer
+- mpeg2 non linear quantizer
+
+unassigned TODO: (unordered)
+---------------
+- use AVFrame for audio codecs too
+- rework aviobuf.c buffering strategy and fix url_fskip
+- generate optimal huffman tables for mjpeg encoding
+- fix ffserver regression tests
+- support xvids motion estimation
+- support x264s motion estimation
+- support x264s rate control
+- SNOW: non translational motion compensation
+- SNOW: more optimal quantization
+- SNOW: 4x4 block support
+- SNOW: 1/8 pel motion compensation support
+- SNOW: iterative motion estimation based on subsampled images
+- SNOW: try B frames and MCTF and see how their PSNR/bitrate/complexity behaves
+- SNOW: try to use the wavelet transformed MC-ed reference frame as context for the entropy coder
+- SNOW: think about/analyize how to make snow use multiple cpus/threads
+- SNOW: finish spec
+- FLAC: lossy encoding (viterbi and naive scalar quantization)
+- libavfilter
+- JPEG2000 decoder & encoder
+- MPEG4 GMC encoding support
+- macroblock based pixel format (better cache locality, somewhat complex, one paper claimed it faster for high res)
+- regression tests for codecs which do not have an encoder (I+P-frame bitstream in the 'master' branch)
+- add support for using mplayers video filters to ffmpeg
+- H264 encoder
+- per MB ratecontrol (so VCD and such do work better)
+- write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions
+- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc
+- generic audio mixing API
+- extract PES packetizer from PS muxer and use it for new TS muxer
+- implement automatic AVBistreamFilter activation
+- make cabac encoder use bytestream (see http://trac.videolan.org/x264/changeset/?format=diff&new=651)
+- merge imdct and windowing, the current code does considerable amounts of redundant work
--- a/doc/authors.texi
+++ b/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.
--- a/doc/avtools-common-opts.texi
+++ b/doc/avtools-common-opts.texi
@@ -1,211 +0,0 @@
-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.
-
-Options which do not take arguments are boolean options, and set the
-corresponding value to true. They can be set to false by prefixing
-with "no" the option name, for example using "-nofoo" in the
-command line will set to false the boolean option with name "foo".
-
-@anchor{Stream specifiers}
-@section Stream specifiers
-Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
-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
-would select the ac3 codec for the second audio stream.
-
-A stream specifier can match several streams, the option is then applied to all
-of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
-streams.
-
-An empty stream specifier matches all streams, for example @code{-codec copy}
-or @code{-codec: copy} would copy all the streams without reencoding.
-
-Possible forms of stream specifiers are:
-@table @option
-@item @var{stream_index}
-Matches the stream with this index. E.g. @code{-threads:1 4} would set the
-thread count for the second stream to 4.
-@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
-'d' for data and 't' for attachments. If @var{stream_index} is given, then
-matches stream number @var{stream_index} of this type. Otherwise matches all
-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.
-
-@table @option
-
-@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 -version
-Show version.
-
-@item -formats
-Show available formats.
-
-The fields preceding the format names have the following meanings:
-@table @samp
-@item D
-Decoding available
-@item E
-Encoding available
-@end table
-
-@item -codecs
-Show all codecs known to libavcodec.
-
-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.
-
-@item -bsfs
-Show available bitstream filters.
-
-@item -protocols
-Show available protocols.
-
-@item -filters
-Show available libavfilter filters.
-
-@item -pix_fmts
-Show available pixel formats.
-
-@item -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:
-@table @samp
-@item quiet
-@item panic
-@item fatal
-@item error
-@item warning
-@item info
-@item verbose
-@item debug
-@end table
-
-By default the program logs to stderr, if coloring is supported by the
-terminal, colors are used to mark errors and warnings. Log coloring
-can be disabled setting the environment variable
-@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
-the environment variable @env{AV_LOG_FORCE_COLOR}.
-The use of the environment variable @env{NO_COLOR} is deprecated and
-will be dropped in a following FFmpeg version.
-
-@item -report
-Dump full command line and console output to a file named
-@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
-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
-
-@end table
-
-@section AVOptions
-
-These options are provided directly by the libavformat, libavdevice and
-libavcodec libraries. To see the list of available AVOptions, use the
-@option{-help} option. They are separated into two categories:
-@table @option
-@item generic
-These options can be set for any container, codec or device. Generic options
-are listed under AVFormatContext options for containers/devices and under
-AVCodecContext options for codecs.
-@item private
-These options are specific to the given container, device or codec. Private
-options are listed under their corresponding containers/devices/codecs.
-@end table
-
-For example to write an ID3v2.3 header instead of a default ID3v2.4 to
-an MP3 file, use the @option{id3v2_version} private option of the MP3
-muxer:
-@example
-ffmpeg -i input.flac -id3v2_version 3 out.mp3
-@end example
-
-All codec AVOptions are obviously per-stream, so the chapter on stream
-specifiers applies to them
-
-Note @option{-nooption} syntax cannot be used for boolean AVOptions,
-use @option{-option 0}/@option{-option 1}.
-
-Note2 old undocumented way of specifying per-stream AVOptions by prepending
-v/a/s to the options name is now obsolete and will be removed soon.
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -23,20 +23,6 @@ Below is a description of the currently available bitstream filters.

@section h264_mp4toannexb

-Convert an H.264 bitstream from length prefixed mode to start code
-prefixed mode (as defined in the Annex B of the ITU-T H.264
-specification).
-
-This is required by some streaming formats, typically the MPEG-2
-transport stream format ("mpegts").
-
-For example to remux an MP4 file containing an H.264 stream to mpegts
-format with @command{ffmpeg}, you can use the command:
-
-@example
-ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
-@end example
-
@section imx_dump_header

@section mjpeg2jpeg
@@ -48,7 +34,7 @@ JPEG image. The individual frames can be extracted without loss,
 e.g. by

@example
-ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
+ffmpeg -i ../some_mjpeg.avi -vcodec copy frames_%d.jpg
@end example

 Unfortunately, these chunks are incomplete JPEG images, because
@@ -71,9 +57,9 @@ 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 -vcodec copy -vbsf mjpeg2jpeg frame_%d.jpg
 exiftran -i -9 frame*.jpg
-ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
+ffmpeg -i frame_%d.jpg -vcodec copy rotated.avi
@end example

@section mjpega_dump_header
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -27,7 +27,7 @@ follows.

@section rawvideo

-Raw video decoder.
+Rawvideo decoder.

 This decoder decodes rawvideo streams.

@@ -48,114 +48,3 @@ top-field-first is assumed
@end table

@c man end VIDEO DECODERS
-
-@chapter Audio Decoders
-@c man begin AUDIO DECODERS
-
-@section ffwavesynth
-
-Internal wave synthetizer.
-
-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
--- a/doc/default.css
+++ b/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;
-}
--- a/doc/demuxers.texi
+++ b/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}, ...,
@@ -67,82 +46,24 @@ 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 @file{ffmpeg} for creating a
+video from the images in the file sequence @file{img-001.jpeg},
+@file{img-002.jpeg}, ..., assuming an input framerate of 10 frames per
+second:
+@example
+ffmpeg -r 10 -f image2 -i 'img-%03d.jpeg' out.avi
+@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:
@example
-ffmpeg -i img.jpeg img.png
+ffmpeg -f image2 -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.
@@ -154,86 +75,4 @@ the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".

-@section sbg
-
-SBaGen script demuxer.
-
-This demuxer reads the script language used by SBaGen
-@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
-script looks like that:
-@example
-SE
-a: 300-2.5/3 440+4.5/0
-b: 300-2.5/0 440+4.5/3
-off: -
-NOW      == a
-+0:07:00 == b
-+0:14:00 == a
-+0:21:00 == b
-+0:30:00    off
-@end example
-
-A SBG script can mix absolute and relative timestamps. If the script uses
-either only absolute timestamps (including the script start time) or only
-relative ones, then its layout is fixed, and the conversion is
-straightforward. On the other hand, if the script mixes both kind of
-timestamps, then the @var{NOW} reference for relative timestamps will be
-taken from the current time of day at the time the script is read, and the
-script layout will be frozen according to that reference. That means that if
-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
--- a/doc/developer.texi
+++ b/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

@@ -35,57 +34,59 @@ You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
 to send your patches to the FFmpeg mailing list.

-@section Contributing
-
-There are 3 ways by which code gets into ffmpeg.
-@itemize @bullet
-@item Submitting Patches to the main developer mailing list
-      see @ref{Submitting patches} for details.
-@item Directly committing changes to the main tree.
-@item Committing changes to a git clone, for example on github.com or
-      gitorious.org. And asking us to merge these changes.
-@end itemize
-
-Whichever way, changes should be reviewed by the maintainer of the code
-before they are committed. And they should follow the @ref{Coding Rules}.
-The developer making the commit and the author are responsible for their changes
-and should try to fix issues their commit causes.
-
@anchor{Coding Rules}
@section Coding Rules

-@subsection Code formatting conventions
-
-There are the following guidelines regarding the indentation in files:
+FFmpeg is programmed in the ISO C90 language with a few additional
+features from ISO C99, namely:
@itemize @bullet
@item
-Indent size is 4.
+the @samp{inline} keyword;
@item
+@samp{//} comments;
+@item
+designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
+@item
+compound literals (@samp{x = (struct s) @{ 17, 23 @};})
+@end itemize
+
+These features are supported by all compilers we care about, so we will not
+accept patches to remove their use unless they absolutely do not impair
+clarity and performance.
+
+All code must compile with GCC 2.95 and GCC 3.3. Currently, FFmpeg also
+compiles with several other compilers, such as the Compaq ccc compiler
+or Sun Studio 9, and we would like to keep it that way unless it would
+be exceedingly involved. To ensure compatibility, please do not use any
+additional C99 features or GCC extensions. Especially watch out for:
+@itemize @bullet
+@item
+mixing statements and declarations;
+@item
+@samp{long long} (use @samp{int64_t} instead);
+@item
+@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
+@item
+GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
+@end itemize
+
+Indent size is 4.
+The presentation is one inspired by 'indent -i4 -kr -nut'.
 The TAB character is forbidden outside of Makefiles as is any
 form of trailing whitespace. Commits containing either will be
 rejected by the git repository.
-@item
-You should try to limit your code lines to 80 characters; however, do so if
-and only if this improves readability.
-@end itemize
-The presentation is one inspired by 'indent -i4 -kr -nut'.

 The main priority in FFmpeg is simplicity and small code size in order to
 minimize the bug count.

-@subsection Comments
-Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
+Comments: Use the JavaDoc/Doxygen
+format (see examples below) so that code documentation
 can be generated automatically. All nontrivial functions should have a comment
 above them explaining what the function does, even if it is just one sentence.
 All structures and their member variables should be documented, too.
-
-Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
-@code{//!} with @code{///} and similar.  Also @@ syntax should be employed
-for markup commands, i.e. use @code{@@param} and not @code{\param}.
-
@example
 /**
- * @@file
+ * @@file mpeg.c
 * MPEG codec.
 * @@author ...
 */
@@ -113,105 +114,11 @@ int myfunc(int my_parameter)
 ...
@end example

-@subsection C language features
-
-FFmpeg is programmed in the ISO C90 language with a few additional
-features from ISO C99, namely:
-@itemize @bullet
-@item
-the @samp{inline} keyword;
-@item
-@samp{//} comments;
-@item
-designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
-@item
-compound literals (@samp{x = (struct s) @{ 17, 23 @};})
-@end itemize
-
-These features are supported by all compilers we care about, so we will not
-accept patches to remove their use unless they absolutely do not impair
-clarity and performance.
-
-All code must compile with recent versions of GCC and a number of other
-currently supported compilers. To ensure compatibility, please do not use
-additional C99 features or GCC extensions. Especially watch out for:
-@itemize @bullet
-@item
-mixing statements and declarations;
-@item
-@samp{long long} (use @samp{int64_t} instead);
-@item
-@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
-@item
-GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
-@end itemize
-
-@subsection Naming conventions
-All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
-a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
-for example structs and enums; they should always be in the CamelCase
-
-
-There are following conventions for naming variables and functions:
-@itemize @bullet
-@item
-For local variables no prefix is required.
-@item
-For variables and functions declared as @code{static} no prefixes are required.
-@item
-For variables and functions used internally by the library, @code{ff_} prefix
-should be used.
-For example, @samp{ff_w64_demuxer}.
-@item
-For variables and functions used internally across multiple libraries, use
-@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
-@item
-For exported names, each library has its own prefixes. Just check the existing
-code and name accordingly.
-@end itemize
-
-@subsection Miscellaneous conventions
-@itemize @bullet
-@item
 fprintf and printf are forbidden in libavformat and libavcodec,
 please use av_log() instead.
-@item
+
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
-@end itemize
-
-@subsection Editor configuration
-In order to configure Vim to follow FFmpeg formatting conventions, paste
-the following snippet into your @file{.vimrc}:
-@example
-" indentation rules for FFmpeg: 4 spaces, no tabs
-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
-" Trailing whitespace and tabs are forbidden, so highlight them.
-highlight ForbiddenWhitespace ctermbg=red guibg=red
-match ForbiddenWhitespace /\s\+$\|\t/
-" Do not highlight spaces at the end of line while typing on that line.
-autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
-@end example
-
-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")
-@end example

@section Development Policy

@@ -273,7 +180,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
   Always fill out the commit log message. Describe in a few lines what you
   changed and why. You can refer to mailing list postings if you fix a
   particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
-   Recommended format:
+   Recommanded format:
   area changed: Short 1 line description

   details describing what and why and giving references.
@@ -308,7 +215,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
    always check values read from some untrusted source before using them
    as array index or other risky things.
@item
-    Remember to check if you need to bump versions for the specific libav*
+    Remember to check if you need to bump versions for the specific libav
    parts (libavutil, libavcodec, libavformat) you are changing. You need
    to change the version integer.
    Incrementing the first component means no backward compatibility to
@@ -317,8 +224,7 @@ 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
@@ -336,11 +242,9 @@ We think our rules are not too hard. If you have comments, contact us.

 Note, these rules are mostly borrowed from the MPlayer project.

-@anchor{Submitting patches}
@section Submitting patches

-First, read the @ref{Coding Rules} above if you did not yet, in particular
-the rules regarding patch submission.
+First, read the (@pxref{Coding Rules}) above if you did not yet.

 When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-)
@@ -355,8 +259,8 @@ for us and greatly increases your chances of getting your patch applied.
 Use the patcheck tool of FFmpeg to check your patch.
 The tool is located in the tools directory.

-Run the @ref{Regression tests} before submitting a patch in order to verify
-it does not cause unexpected problems.
+Run the regression tests before submitting a patch so that you can
+verify that there are no big problems.

 Patches should be posted as base64 encoded attachments (or any other
 encoding which ensures that the patch will not be trashed during
@@ -390,15 +294,13 @@ send a reminder by email. Your patch should eventually be dealt with.
    AVInputFormat/AVOutputFormat struct?
@item
    Did you bump the minor version number (and reset the micro version
-    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
+    number) in @file{avcodec.h} or @file{avformat.h}?
@item
    Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
-    Did you add the 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},
+    If it has a fourcc, did you add it to @file{libavformat/riff.c},
    even if it is only a decoder?
@item
    Did you add a rule to compile the appropriate files in the Makefile?
@@ -425,19 +327,19 @@ send a reminder by email. Your patch should eventually be dealt with.

@enumerate
@item
-    Does @code{make fate} pass with the patch applied?
+    Does 'make fate' pass with the patch applied?
@item
    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?
@item
    Is the patch against latest FFmpeg git master branch?
@item
-    Are you subscribed to ffmpeg-devel?
+    Are you subscribed to ffmpeg-dev?
    (the list is subscribers only due to spam)
@item
    Have you checked that the changes are minimal, so that the same cannot be
@@ -492,10 +394,6 @@ 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
@@ -514,38 +412,37 @@ After a patch is approved it will be committed to the repository.
 We will review all submitted patches, but sometimes we are quite busy so
 especially for large patches this can take several weeks.

-If you feel that the review process is too slow and you are willing to try to
-take over maintainership of the area of code you change then just clone
-git master and maintain the area of code there. We will merge each area from
-where its best maintained.
-
 When resubmitting patches, please do not make any significant changes
 not related to the comments received during review. Such patches will
-be rejected. Instead, submit significant changes or new features as
+be rejected. Instead, submit  significant changes or new features as
 separate patches.

-@anchor{Regression tests}
@section Regression tests

 Before submitting a patch (or committing to the repository), you should at least
 test that you did not break anything.

-Running 'make fate' accomplishes this, please see @url{fate.html} for details.
+The regression tests build a synthetic video stream and a synthetic
+audio stream. These are then encoded and decoded with all codecs or
+formats. The CRC (or MD5) of each generated file is recorded in a
+result file. A 'diff' is launched to compare the reference results and
+the result file. The output is checked immediately after each test
+has run.
+
+The regression tests then go on to test the FFserver code with a
+limited set of streams. It is important that this step runs correctly
+as well.
+
+Run 'make test' to test all the codecs and formats. Commands like
+'make regtest-mpeg2' can be used to run a single test. By default,
+make will abort if any test fails. To run all tests regardless,
+use make -k. To get a more verbose output, use 'make V=1 test' or
+'make V=2 test'.
+
+Run 'make fulltest' to test all the codecs, formats and FFserver.

 [Of course, some patches may change the results of the regression tests. In
 this case, the reference results of the regression tests shall be modified
 accordingly].

-@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
--- a/doc/doxy-wrapper.sh
+++ b/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
--- a/doc/doxy/doxy_stylesheet.css
+++ b/doc/doxy/doxy_stylesheet.css
--- a/doc/doxy/footer.html
+++ b/doc/doxy/footer.html
@@ -1,9 +0,0 @@
-
- <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> $doxygenversion
-   </span>
- </footer>
-</div>
-</body>
-</html>
--- a/doc/doxy/header.html
+++ b/doc/doxy/header.html
@@ -1,16 +0,0 @@
-<!DOCTYPE html>
-<html>
-<head>
-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
-<meta http-equiv="X-UA-Compatible" content="IE=9"/>
-<!--BEGIN PROJECT_NAME--><title>$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">
-
-<!--Header replace -->
-<div class="menu">
--- a/doc/encoders.texi
+++ b/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.
@@ -409,10 +320,10 @@ apply Dolby Surround EX processing.
 Not Indicated (default)
@item 1
@itemx on
-Dolby Surround EX Off
+Dolby Surround EX On
@item 2
@itemx off
-Dolby Surround EX On
+Dolby Surround EX Off
@end table

@item -dheadphone_mode @var{mode}
@@ -426,10 +337,10 @@ processing.
 Not Indicated (default)
@item 1
@itemx on
-Dolby Headphone Off
+Dolby Headphone On
@item 2
@itemx off
-Dolby Headphone On
+Dolby Headphone Off
@end table

@item -ad_conv_type @var{type}
@@ -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
@@ -959,35 +551,36 @@ Set the encoding preset.

@item tune @var{tune_name}
 Tune the encoding params.
+Deprecated in favor of @var{x264_opts}

@item fastfirstpass @var{bool}
 Use fast settings when encoding first pass, default value is 1.
+Deprecated in favor of @var{x264_opts}.

@item profile @var{profile_name}
 Set profile restrictions.
+Deprecated in favor of @var{x264_opts}.

@item level @var{level}
 Specify level (as defined by Annex A).
-Deprecated in favor of @var{x264opts}.
+Deprecated in favor of @var{x264_opts}.

@item passlogfile @var{filename}
 Specify filename for 2 pass stats.
-Deprecated in favor of @var{x264opts} (see @var{stats} libx264 option).
+Deprecated in favor of @var{x264_opts}.

@item wpredp @var{wpred_type}
 Specify Weighted prediction for P-frames.
-Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).
+Deprecated in favor of @var{x264_opts}.

@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 manual 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}:
+For example to specify libx264 encoding options with @file{ffmpeg}:
@example
 ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example
@@ -995,117 +588,4 @@ 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
--- a/doc/errno.txt
+++ b/doc/errno.txt
@@ -1,174 +0,0 @@
-The following table lists most error codes found in various operating
-systems supported by FFmpeg.
-
-                             OS
-Code             Std    F  LBMWwb Text (YMMV)
-
-E2BIG            POSIX     ++++++  Argument list too long
-EACCES           POSIX     ++++++  Permission denied
-EADDRINUSE       POSIX     +++..+  Address in use
-EADDRNOTAVAIL    POSIX     +++..+  Cannot assign requested address
-EADV                       +.....  Advertise error
-EAFNOSUPPORT     POSIX     +++..+  Address family not supported
-EAGAIN           POSIX  +  ++++++  Resource temporarily unavailable
-EALREADY         POSIX     +++..+  Operation already in progress
-EAUTH                      .++...  Authentication error
-EBADARCH                   ..+...  Bad CPU type in executable
-EBADE                      +.....  Invalid exchange
-EBADEXEC                   ..+...  Bad executable
-EBADF            POSIX     ++++++  Bad file descriptor
-EBADFD                     +.....  File descriptor in bad state
-EBADMACHO                  ..+...  Malformed Macho file
-EBADMSG          POSIX     ++4...  Bad message
-EBADR                      +.....  Invalid request descriptor
-EBADRPC                    .++...  RPC struct is bad
-EBADRQC                    +.....  Invalid request code
-EBADSLT                    +.....  Invalid slot
-EBFONT                     +.....  Bad font file format
-EBUSY            POSIX  -  ++++++  Device or resource busy
-ECANCELED        POSIX     +++...  Operation canceled
-ECHILD           POSIX     ++++++  No child processes
-ECHRNG                     +.....  Channel number out of range
-ECOMM                      +.....  Communication error on send
-ECONNABORTED     POSIX     +++..+  Software caused connection abort
-ECONNREFUSED     POSIX  -  +++ss+  Connection refused
-ECONNRESET       POSIX     +++..+  Connection reset
-EDEADLK          POSIX     ++++++  Resource deadlock avoided
-EDEADLOCK                  +..++.  File locking deadlock error
-EDESTADDRREQ     POSIX     +++...  Destination address required
-EDEVERR                    ..+...  Device error
-EDOM             C89    -  ++++++  Numerical argument out of domain
-EDOOFUS                    .F....  Programming error
-EDOTDOT                    +.....  RFS specific error
-EDQUOT           POSIX     +++...  Disc quota exceeded
-EEXIST           POSIX     ++++++  File exists
-EFAULT           POSIX  -  ++++++  Bad address
-EFBIG            POSIX  -  ++++++  File too large
-EFTYPE                     .++...  Inappropriate file type or format
-EHOSTDOWN                  +++...  Host is down
-EHOSTUNREACH     POSIX     +++..+  No route to host
-EHWPOISON                  +.....  Memory page has hardware error
-EIDRM            POSIX     +++...  Identifier removed
-EILSEQ           C99       ++++++  Illegal byte sequence
-EINPROGRESS      POSIX  -  +++ss+  Operation in progress
-EINTR            POSIX  -  ++++++  Interrupted system call
-EINVAL           POSIX  +  ++++++  Invalid argument
-EIO              POSIX  +  ++++++  I/O error
-EISCONN          POSIX     +++..+  Socket is already connected
-EISDIR           POSIX     ++++++  Is a directory
-EISNAM                     +.....  Is a named type file
-EKEYEXPIRED                +.....  Key has expired
-EKEYREJECTED               +.....  Key was rejected by service
-EKEYREVOKED                +.....  Key has been revoked
-EL2HLT                     +.....  Level 2 halted
-EL2NSYNC                   +.....  Level 2 not synchronized
-EL3HLT                     +.....  Level 3 halted
-EL3RST                     +.....  Level 3 reset
-ELIBACC                    +.....  Can not access a needed shared library
-ELIBBAD                    +.....  Accessing a corrupted shared library
-ELIBEXEC                   +.....  Cannot exec a shared library directly
-ELIBMAX                    +.....  Too many shared libraries
-ELIBSCN                    +.....  .lib section in a.out corrupted
-ELNRNG                     +.....  Link number out of range
-ELOOP            POSIX     +++..+  Too many levels of symbolic links
-EMEDIUMTYPE                +.....  Wrong medium type
-EMFILE           POSIX     ++++++  Too many open files
-EMLINK           POSIX     ++++++  Too many links
-EMSGSIZE         POSIX     +++..+  Message too long
-EMULTIHOP        POSIX     ++4...  Multihop attempted
-ENAMETOOLONG     POSIX  -  ++++++  Filen ame too long
-ENAVAIL                    +.....  No XENIX semaphores available
-ENEEDAUTH                  .++...  Need authenticator
-ENETDOWN         POSIX     +++..+  Network is down
-ENETRESET        SUSv3     +++..+  Network dropped connection on reset
-ENETUNREACH      POSIX     +++..+  Network unreachable
-ENFILE           POSIX     ++++++  Too many open files in system
-ENOANO                     +.....  No anode
-ENOATTR                    .++...  Attribute not found
-ENOBUFS          POSIX  -  +++..+  No buffer space available
-ENOCSI                     +.....  No CSI structure available
-ENODATA          XSR       +N4...  No message available
-ENODEV           POSIX  -  ++++++  No such device
-ENOENT           POSIX  -  ++++++  No such file or directory
-ENOEXEC          POSIX     ++++++  Exec format error
-ENOFILE                    ...++.  No such file or directory
-ENOKEY                     +.....  Required key not available
-ENOLCK           POSIX     ++++++  No locks available
-ENOLINK          POSIX     ++4...  Link has been severed
-ENOMEDIUM                  +.....  No medium found
-ENOMEM           POSIX     ++++++  Not enough space
-ENOMSG           POSIX     +++..+  No message of desired type
-ENONET                     +.....  Machine is not on the network
-ENOPKG                     +.....  Package not installed
-ENOPROTOOPT      POSIX     +++..+  Protocol not available
-ENOSPC           POSIX     ++++++  No space left on device
-ENOSR            XSR       +N4...  No STREAM resources
-ENOSTR           XSR       +N4...  Not a STREAM
-ENOSYS           POSIX  +  ++++++  Function not implemented
-ENOTBLK                    +++...  Block device required
-ENOTCONN         POSIX     +++..+  Socket is not connected
-ENOTDIR          POSIX     ++++++  Not a directory
-ENOTEMPTY        POSIX     ++++++  Directory not empty
-ENOTNAM                    +.....  Not a XENIX named type file
-ENOTRECOVERABLE  SUSv4  -  +.....  State not recoverable
-ENOTSOCK         POSIX     +++..+  Socket operation on non-socket
-ENOTSUP          POSIX     +++...  Operation not supported
-ENOTTY           POSIX     ++++++  Inappropriate I/O control operation
-ENOTUNIQ                   +.....  Name not unique on network
-ENXIO            POSIX     ++++++  No such device or address
-EOPNOTSUPP       POSIX     +++..+  Operation not supported (on socket)
-EOVERFLOW        POSIX     +++..+  Value too large to be stored in data type
-EOWNERDEAD       SUSv4     +.....  Owner died
-EPERM            POSIX  -  ++++++  Operation not permitted
-EPFNOSUPPORT               +++..+  Protocol family not supported
-EPIPE            POSIX  -  ++++++  Broken pipe
-EPROCLIM                   .++...  Too many processes
-EPROCUNAVAIL               .++...  Bad procedure for program
-EPROGMISMATCH              .++...  Program version wrong
-EPROGUNAVAIL               .++...  RPC prog. not avail
-EPROTO           POSIX     ++4...  Protocol error
-EPROTONOSUPPORT  POSIX  -  +++ss+  Protocol not supported
-EPROTOTYPE       POSIX     +++..+  Protocol wrong type for socket
-EPWROFF                    ..+...  Device power is off
-ERANGE           C89    -  ++++++  Result too large
-EREMCHG                    +.....  Remote address changed
-EREMOTE                    +++...  Object is remote
-EREMOTEIO                  +.....  Remote I/O error
-ERESTART                   +.....  Interrupted system call should be restarted
-ERFKILL                    +.....  Operation not possible due to RF-kill
-EROFS            POSIX     ++++++  Read-only file system
-ERPCMISMATCH               .++...  RPC version wrong
-ESHLIBVERS                 ..+...  Shared library version mismatch
-ESHUTDOWN                  +++..+  Cannot send after socket shutdown
-ESOCKTNOSUPPORT            +++...  Socket type not supported
-ESPIPE           POSIX     ++++++  Illegal seek
-ESRCH            POSIX     ++++++  No such process
-ESRMNT                     +.....  Srmount error
-ESTALE           POSIX     +++..+  Stale NFS file handle
-ESTRPIPE                   +.....  Streams pipe error
-ETIME            XSR       +N4...  Stream ioctl timeout
-ETIMEDOUT        POSIX  -  +++ss+  Connection timed out
-ETOOMANYREFS               +++...  Too many references: cannot splice
-ETXTBSY          POSIX     +++...  Text file busy
-EUCLEAN                    +.....  Structure needs cleaning
-EUNATCH                    +.....  Protocol driver not attached
-EUSERS                     +++...  Too many users
-EWOULDBLOCK      POSIX     +++..+  Operation would block
-EXDEV            POSIX     ++++++  Cross-device link
-EXFULL                     +.....  Exchange full
-
-Notations:
-
-F: used in FFmpeg (-: a few times, +: a lot)
-
-SUSv3: Single Unix Specification, version 3
-SUSv4: Single Unix Specification, version 4
-XSR: XSI STREAMS (obsolete)
-
-OS: availability on some supported operating systems
-L: GNU/Linux
-B: BSD (F: FreeBSD, N: NetBSD)
-M: MacOS X
-W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h)
-w: Mingw32 (3.17) and Mingw64 (2.0.1)
-b: BeOS
--- a/doc/eval.texi
+++ b/doc/eval.texi
@@ -1,7 +1,7 @@
@chapter Expression Evaluation
@c man begin EXPRESSION EVALUATION

-When evaluating an arithmetic expression, FFmpeg uses an internal
+When evaluating an arithemetic expression, FFmpeg uses an internal
 formula evaluator, implemented through the @file{libavutil/eval.h}
 interface.

@@ -21,88 +21,39 @@ 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
 store the value, and it is a value ranging from 0 to 9. The function
 returns the value stored in the internal variable.
-Note, Variables are currently not shared between expressions.

@item ld(var)
 Allow to load the value of the internal variable with number
-@var{var}, which was previously stored with st(@var{var}, @var{expr}).
+@var{var}, which was previosly stored with st(@var{var}, @var{expr}).
 The function returns the loaded value.

@item while(cond, expr)
@@ -132,66 +83,21 @@ Return 1.0 if @var{expr} is zero, 0.0 otherwise.
@item pow(x, y)
 Compute the power of @var{x} elevated @var{y}, it is equivalent to
 "(@var{x})^(@var{y})".
-
-@item random(x)
-Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
-internal variable which will be used to save the seed/state.
-
-@item hypot(x, y)
-This function is similar to the C function with the same name; it returns
-"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
-right triangle with sides of length @var{x} and @var{y}, or the distance of the
-point (@var{x}, @var{y}) from the origin.
-
-@item gcd(x, y)
-Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
-@var{y} are 0 or either or both are less than zero then behavior is undefined.
-
-@item if(x, y)
-Evaluate @var{x}, and if the result is non-zero return the result of
-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:
-@table @option
-@item PI
-area of the unit disc, approximately 3.14
-@item E
-exp(1) (Euler's number), approximately 2.718
-@item PHI
-golden ratio (1+sqrt(5))/2, approximately 1.618
-@end table
-
-Assuming that an expression is considered "true" if it has a non-zero
-value, note that:
+Note that:

@code{*} works like AND

@code{+} works like OR

-and the construct:
+thus
@example
 if A then B else C
@end example
 is equivalent to
@example
-if(A,B) + ifnot(A,C)
+A*B + not(A)*C
@end example

 In your C code, you can extend the list of unary and binary functions,
--- a/doc/examples/Makefile
+++ b/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=encoding-example muxing-example

 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)
--- a/doc/examples/README
+++ b/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.
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -1,650 +0,0 @@
-/*
- * Copyright (c) 2001 Fabrice Bellard
- *
- * 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
- * libavcodec API use example.
- *
- * 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>
-
-#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
- */
-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;
-    FILE *f;
-    uint16_t *samples;
-    float t, tincr;
-
-    printf("Encode audio file %s\n", filename);
-
-    /* find the MP2 encoder */
-    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
-    if (!codec) {
-        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_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");
-        exit(1);
-    }
-
-    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");
-        exit(1);
-    }
-
-    /* encode a single tone sound */
-    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++) {
-            samples[2*j] = (int)(sin(t) * 10000);
-
-            for (k = 1; k < c->channels; k++)
-                samples[2*j + k] = 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);
-        }
-    }
-    fclose(f);
-
-    av_freep(&samples);
-    avcodec_free_frame(&frame);
-    avcodec_close(c);
-    av_free(c);
-}
-
-/*
- * Audio decoding.
- */
-static void audio_decode_example(const char *outfilename, const char *filename)
-{
-    AVCodec *codec;
-    AVCodecContext *c= NULL;
-    int len;
-    FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
-    AVPacket avpkt;
-    AVFrame *decoded_frame = NULL;
-
-    av_init_packet(&avpkt);
-
-    printf("Decode audio file %s to %s\n", filename, outfilename);
-
-    /* find the mpeg audio decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
-    if (!codec) {
-        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");
-        exit(1);
-    }
-
-    f = fopen(filename, "rb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-    outfile = fopen(outfilename, "wb");
-    if (!outfile) {
-        av_free(c);
-        exit(1);
-    }
-
-    /* decode until eof */
-    avpkt.data = inbuf;
-    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
-
-    while (avpkt.size > 0) {
-        int got_frame = 0;
-
-        if (!decoded_frame) {
-            if (!(decoded_frame = avcodec_alloc_frame())) {
-                fprintf(stderr, "Could not allocate audio frame\n");
-                exit(1);
-            }
-        } else
-            avcodec_get_frame_defaults(decoded_frame);
-
-        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
-        if (len < 0) {
-            fprintf(stderr, "Error while decoding\n");
-            exit(1);
-        }
-        if (got_frame) {
-            /* if a frame has been decoded, output it */
-            int data_size = av_samples_get_buffer_size(NULL, c->channels,
-                                                       decoded_frame->nb_samples,
-                                                       c->sample_fmt, 1);
-            fwrite(decoded_frame->data[0], 1, data_size, outfile);
-        }
-        avpkt.size -= len;
-        avpkt.data += len;
-        avpkt.dts =
-        avpkt.pts = AV_NOPTS_VALUE;
-        if (avpkt.size < AUDIO_REFILL_THRESH) {
-            /* Refill the input buffer, to avoid trying to decode
-             * incomplete frames. Instead of this, one could also use
-             * a parser, or use a proper container format through
-             * libavformat. */
-            memmove(inbuf, avpkt.data, avpkt.size);
-            avpkt.data = inbuf;
-            len = fread(avpkt.data + avpkt.size, 1,
-                        AUDIO_INBUF_SIZE - avpkt.size, f);
-            if (len > 0)
-                avpkt.size += len;
-        }
-    }
-
-    fclose(outfile);
-    fclose(f);
-
-    avcodec_close(c);
-    av_free(c);
-    avcodec_free_frame(&decoded_frame);
-}
-
-/*
- * Video encoding example
- */
-static void video_encode_example(const char *filename, int codec_id)
-{
-    AVCodec *codec;
-    AVCodecContext *c= NULL;
-    int i, ret, x, y, got_output;
-    FILE *f;
-    AVFrame *frame;
-    AVPacket pkt;
-    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
-
-    printf("Encode video file %s\n", filename);
-
-    /* find the mpeg1 video encoder */
-    codec = avcodec_find_encoder(codec_id);
-    if (!codec) {
-        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);
-    }
-
-    /* put sample parameters */
-    c->bit_rate = 400000;
-    /* resolution must be a multiple of two */
-    c->width = 352;
-    c->height = 288;
-    /* frames per second */
-    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;
-
-    if(codec_id == AV_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");
-        exit(1);
-    }
-
-    f = fopen(filename, "wb");
-    if (!f) {
-        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;
-
-    /* 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);
-    }
-
-    /* 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;
-            }
-        }
-
-        /* 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;
-            }
-        }
-
-        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);
-        }
-    }
-
-    /* get the delayed frames */
-    for (got_output = 1; got_output; 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);
-        }
-    }
-
-    /* add sequence end code to have a real mpeg file */
-    fwrite(endcode, 1, sizeof(endcode), f);
-    fclose(f);
-
-    avcodec_close(c);
-    av_free(c);
-    av_freep(&frame->data[0]);
-    avcodec_free_frame(&frame);
-    printf("\n");
-}
-
-/*
- * Video decoding example
- */
-
-static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
-                     char *filename)
-{
-    FILE *f;
-    int i;
-
-    f=fopen(filename,"w");
-    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
-    for(i=0;i<ysize;i++)
-        fwrite(buf + i * wrap,1,xsize,f);
-    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;
-    FILE *f;
-    AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
-    AVPacket avpkt;
-
-    av_init_packet(&avpkt);
-
-    /* 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);
-
-    /* find the mpeg1 video decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
-    if (!codec) {
-        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);
-    }
-
-    if(codec->capabilities&CODEC_CAP_TRUNCATED)
-        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
-
-    /* For some codecs, such as msmpeg4 and mpeg4, width and height
-       MUST be initialized there because this information is not
-       available in the bitstream. */
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "rb");
-    if (!f) {
-        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;
-    for(;;) {
-        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
-        if (avpkt.size == 0)
-            break;
-
-        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
-           and this is the only method to use them because you cannot
-           know the compressed data size before analysing it.
-
-           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
-           based, so you must call them with all the data for one
-           frame exactly. You must also initialize 'width' and
-           'height' before initializing them. */
-
-        /* NOTE2: some codecs allow the raw parameters (frame size,
-           sample rate) to be changed at any frame. We handle this, so
-           you should also take care of it */
-
-        /* 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)
-                exit(1);
-    }
-
-    /* some codecs, such as MPEG, transmit the I and P frame with a
-       latency of one frame. You must do the following to have a
-       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);
-
-    fclose(f);
-
-    avcodec_close(c);
-    av_free(c);
-    avcodec_free_frame(&frame);
-    printf("\n");
-}
-
-int main(int argc, char **argv)
-{
-    const char *output_type;
-
-    /* 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 (!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");
-    } else {
-        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
-                output_type);
-        return 1;
-    }
-
-    return 0;
-}
--- a/doc/examples/demuxing.c
+++ b/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;
-}
--- a/doc/examples/encoding-example.c
+++ b/doc/examples/encoding-example.c
@@ -0,0 +1,467 @@
+/*
+ * Copyright (c) 2001 Fabrice Bellard
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * avcodec API use example.
+ *
+ * Note that this library only handles codecs (mpeg, mpeg4, etc...),
+ * not file formats (avi, vob, etc...). See library 'libavformat' for the
+ * format handling
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#ifdef HAVE_AV_CONFIG_H
+#undef HAVE_AV_CONFIG_H
+#endif
+
+#include "libavcodec/avcodec.h"
+#include "libavutil/mathematics.h"
+
+#define INBUF_SIZE 4096
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
+
+/*
+ * Audio encoding example
+ */
+static void audio_encode_example(const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int frame_size, i, j, out_size, outbuf_size;
+    FILE *f;
+    short *samples;
+    float t, tincr;
+    uint8_t *outbuf;
+
+    printf("Audio encoding\n");
+
+    /* find the MP2 encoder */
+    codec = avcodec_find_encoder(CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c= avcodec_alloc_context();
+
+    /* put sample parameters */
+    c->bit_rate = 64000;
+    c->sample_rate = 44100;
+    c->channels = 2;
+
+    /* open it */
+    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);
+    }
+
+    /* encode a single tone sound */
+    t = 0;
+    tincr = 2 * M_PI * 440.0 / c->sample_rate;
+    for(i=0;i<200;i++) {
+        for(j=0;j<frame_size;j++) {
+            samples[2*j] = (int)(sin(t) * 10000);
+            samples[2*j+1] = samples[2*j];
+            t += tincr;
+        }
+        /* encode the samples */
+        out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
+        fwrite(outbuf, 1, out_size, f);
+    }
+    fclose(f);
+    free(outbuf);
+    free(samples);
+
+    avcodec_close(c);
+    av_free(c);
+}
+
+/*
+ * Audio decoding.
+ */
+static void audio_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int out_size, len;
+    FILE *f, *outfile;
+    uint8_t *outbuf;
+    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+
+    av_init_packet(&avpkt);
+
+    printf("Audio decoding\n");
+
+    /* find the mpeg audio decoder */
+    codec = avcodec_find_decoder(CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c= avcodec_alloc_context();
+
+    /* open it */
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
+        exit(1);
+    }
+
+    outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "could not open %s\n", filename);
+        exit(1);
+    }
+    outfile = fopen(outfilename, "wb");
+    if (!outfile) {
+        av_free(c);
+        exit(1);
+    }
+
+    /* decode until eof */
+    avpkt.data = inbuf;
+    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+    while (avpkt.size > 0) {
+        out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+        len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt);
+        if (len < 0) {
+            fprintf(stderr, "Error while decoding\n");
+            exit(1);
+        }
+        if (out_size > 0) {
+            /* if a frame has been decoded, output it */
+            fwrite(outbuf, 1, out_size, outfile);
+        }
+        avpkt.size -= len;
+        avpkt.data += len;
+        if (avpkt.size < AUDIO_REFILL_THRESH) {
+            /* Refill the input buffer, to avoid trying to decode
+             * incomplete frames. Instead of this, one could also use
+             * a parser, or use a proper container format through
+             * libavformat. */
+            memmove(inbuf, avpkt.data, avpkt.size);
+            avpkt.data = inbuf;
+            len = fread(avpkt.data + avpkt.size, 1,
+                        AUDIO_INBUF_SIZE - avpkt.size, f);
+            if (len > 0)
+                avpkt.size += len;
+        }
+    }
+
+    fclose(outfile);
+    fclose(f);
+    free(outbuf);
+
+    avcodec_close(c);
+    av_free(c);
+}
+
+/*
+ * Video encoding example
+ */
+static void video_encode_example(const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int i, out_size, size, x, y, outbuf_size;
+    FILE *f;
+    AVFrame *picture;
+    uint8_t *outbuf, *picture_buf;
+
+    printf("Video encoding\n");
+
+    /* find the mpeg1 video encoder */
+    codec = avcodec_find_encoder(CODEC_ID_MPEG1VIDEO);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c= avcodec_alloc_context();
+    picture= avcodec_alloc_frame();
+
+    /* put sample parameters */
+    c->bit_rate = 400000;
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* frames per second */
+    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 = PIX_FMT_YUV420P;
+
+    /* open it */
+    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);
+        exit(1);
+    }
+
+    /* alloc image and output buffer */
+    outbuf_size = 100000;
+    outbuf = malloc(outbuf_size);
+    size = c->width * c->height;
+    picture_buf = malloc((size * 3) / 2); /* size for YUV 420 */
+
+    picture->data[0] = picture_buf;
+    picture->data[1] = picture->data[0] + size;
+    picture->data[2] = picture->data[1] + size / 4;
+    picture->linesize[0] = c->width;
+    picture->linesize[1] = c->width / 2;
+    picture->linesize[2] = c->width / 2;
+
+    /* encode 1 second of video */
+    for(i=0;i<25;i++) {
+        fflush(stdout);
+        /* prepare a dummy image */
+        /* Y */
+        for(y=0;y<c->height;y++) {
+            for(x=0;x<c->width;x++) {
+                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++) {
+                picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
+                picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
+            }
+        }
+
+        /* encode the image */
+        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(; out_size; i++) {
+        fflush(stdout);
+
+        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 */
+    outbuf[0] = 0x00;
+    outbuf[1] = 0x00;
+    outbuf[2] = 0x01;
+    outbuf[3] = 0xb7;
+    fwrite(outbuf, 1, 4, f);
+    fclose(f);
+    free(picture_buf);
+    free(outbuf);
+
+    avcodec_close(c);
+    av_free(c);
+    av_free(picture);
+    printf("\n");
+}
+
+/*
+ * Video decoding example
+ */
+
+static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
+                     char *filename)
+{
+    FILE *f;
+    int i;
+
+    f=fopen(filename,"w");
+    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
+    for(i=0;i<ysize;i++)
+        fwrite(buf + i * wrap,1,xsize,f);
+    fclose(f);
+}
+
+static void video_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int frame, got_picture, len;
+    FILE *f;
+    AVFrame *picture;
+    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    char buf[1024];
+    AVPacket avpkt;
+
+    av_init_packet(&avpkt);
+
+    /* 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("Video decoding\n");
+
+    /* find the mpeg1 video decoder */
+    codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c= avcodec_alloc_context();
+    picture= avcodec_alloc_frame();
+
+    if(codec->capabilities&CODEC_CAP_TRUNCATED)
+        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
+
+    /* For some codecs, such as msmpeg4 and mpeg4, width and height
+       MUST be initialized there because this information is not
+       available in the bitstream. */
+
+    /* open it */
+    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);
+        exit(1);
+    }
+
+    frame = 0;
+    for(;;) {
+        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
+        if (avpkt.size == 0)
+            break;
+
+        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
+           and this is the only method to use them because you cannot
+           know the compressed data size before analysing it.
+
+           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
+           based, so you must call them with all the data for one
+           frame exactly. You must also initialize 'width' and
+           'height' before initializing them. */
+
+        /* NOTE2: some codecs allow the raw parameters (frame size,
+           sample rate) to be changed at any frame. We handle this, so
+           you should also take care of it */
+
+        /* 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) {
+            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
+       latency of one frame. You must do the following to have a
+       chance to get the last frame of the video */
+    avpkt.data = NULL;
+    avpkt.size = 0;
+    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);
+    av_free(picture);
+    printf("\n");
+}
+
+int main(int argc, char **argv)
+{
+    const char *filename;
+
+    /* must be called before using avcodec lib */
+    avcodec_init();
+
+    /* register all the codecs */
+    avcodec_register_all();
+
+    if (argc <= 1) {
+        audio_encode_example("/tmp/test.mp2");
+        audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
+
+        video_encode_example("/tmp/test.mpg");
+        filename = "/tmp/test.mpg";
+    } else {
+        filename = argv[1];
+    }
+
+    //    audio_decode_example("/tmp/test.sw", filename);
+    video_decode_example("/tmp/test%d.pgm", filename);
+
+    return 0;
+}
--- a/doc/examples/filtering_audio.c
+++ b/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);
-}
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -1,248 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 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
- * 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>
-
-const char *filter_descr = "scale=78:24";
-
-static AVFormatContext *fmt_ctx;
-static AVCodecContext *dec_ctx;
-AVFilterContext *buffersink_ctx;
-AVFilterContext *buffersrc_ctx;
-AVFilterGraph *filter_graph;
-static int video_stream_index = -1;
-static int64_t last_pts = AV_NOPTS_VALUE;
-
-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 video stream */
-    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
-        return ret;
-    }
-    video_stream_index = ret;
-    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
-
-    /* init the video decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-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");
-    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;
-
-    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);
-
-    ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                                       args, NULL, filter_graph);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        return ret;
-    }
-
-    /* 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);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create 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;
-    return 0;
-}
-
-static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
-{
-    int x, y;
-    uint8_t *p0, *p;
-    int64_t delay;
-
-    if (picref->pts != AV_NOPTS_VALUE) {
-        if (last_pts != AV_NOPTS_VALUE) {
-            /* sleep roughly the right amount of time;
-             * usleep is in microseconds, just like AV_TIME_BASE. */
-            delay = av_rescale_q(picref->pts - last_pts,
-                                 time_base, AV_TIME_BASE_Q);
-            if (delay > 0 && delay < 1000000)
-                usleep(delay);
-        }
-        last_pts = picref->pts;
-    }
-
-    /* Trivial ASCII grayscale display. */
-    p0 = picref->data[0];
-    puts("\033c");
-    for (y = 0; y < picref->video->h; y++) {
-        p = p0;
-        for (x = 0; x < picref->video->w; x++)
-            putchar(" .-+#"[*(p++) / 52]);
-        putchar('\n');
-        p0 += picref->linesize[0];
-    }
-    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\n", argv[0]);
-        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 *picref;
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
-
-        if (packet.stream_index == video_stream_index) {
-            avcodec_get_frame_defaults(&frame);
-            got_frame = 0;
-            ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &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);
-
-                /* 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;
-                }
-
-                /* 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;
-
-                    if (picref) {
-                        display_picref(picref, buffersink_ctx->inputs[0]->time_base);
-                        avfilter_unref_bufferp(&picref);
-                    }
-                }
-            }
-        }
-        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);
-}
--- a/doc/examples/metadata.c
+++ b/doc/examples/metadata.c
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 Reinhard Tartler
- *
- * 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
- * Shows how the metadata API can be used in application programs.
- * @example doc/examples/metadata.c
- */
-
-#include <stdio.h>
-
-#include <libavformat/avformat.h>
-#include <libavutil/dict.h>
-
-int main (int argc, char **argv)
-{
-    AVFormatContext *fmt_ctx = NULL;
-    AVDictionaryEntry *tag = NULL;
-    int ret;
-
-    if (argc != 2) {
-        printf("usage: %s <input_file>\n"
-               "example program to demonstrate the use of the libavformat metadata API.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    av_register_all();
-    if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
-        return ret;
-
-    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);
-    return 0;
-}
--- a/doc/examples/muxing-example.c
+++ b/doc/examples/muxing-example.c
@@ -0,0 +1,542 @@
+/*
+ * Copyright (c) 2003 Fabrice Bellard
+ *
+ * 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 API example: Output a media file in any supported
+ * libavformat format. The default codecs are used.
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+#include "libavformat/avformat.h"
+#include "libswscale/swscale.h"
+
+#undef exit
+
+/* 5 seconds stream duration */
+#define STREAM_DURATION   5.0
+#define STREAM_FRAME_RATE 25 /* 25 images/s */
+#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
+#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */
+
+static int sws_flags = SWS_BICUBIC;
+
+/**************************************************************/
+/* audio output */
+
+float t, tincr, tincr2;
+int16_t *samples;
+uint8_t *audio_outbuf;
+int audio_outbuf_size;
+int audio_input_frame_size;
+
+/*
+ * add an audio output stream
+ */
+static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
+{
+    AVCodecContext *c;
+    AVStream *st;
+
+    st = av_new_stream(oc, 1);
+    if (!st) {
+        fprintf(stderr, "Could not alloc stream\n");
+        exit(1);
+    }
+
+    c = st->codec;
+    c->codec_id = codec_id;
+    c->codec_type = AVMEDIA_TYPE_AUDIO;
+
+    /* put sample parameters */
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    c->bit_rate = 64000;
+    c->sample_rate = 44100;
+    c->channels = 2;
+
+    // some formats want stream headers to be separate
+    if(oc->oformat->flags & AVFMT_GLOBALHEADER)
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    return st;
+}
+
+static void open_audio(AVFormatContext *oc, AVStream *st)
+{
+    AVCodecContext *c;
+    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 */
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
+        exit(1);
+    }
+
+    /* init signal generator */
+    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;
+
+    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 * 2 * c->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;
+    int16_t *q;
+
+    q = samples;
+    for(j=0;j<frame_size;j++) {
+        v = (int)(sin(t) * 10000);
+        for(i = 0; i < nb_channels; i++)
+            *q++ = v;
+        t += tincr;
+        tincr += tincr2;
+    }
+}
+
+static void write_audio_frame(AVFormatContext *oc, AVStream *st)
+{
+    AVCodecContext *c;
+    AVPacket pkt;
+    av_init_packet(&pkt);
+
+    c = st->codec;
+
+    get_audio_frame(samples, audio_input_frame_size, c->channels);
+
+    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 in the media file */
+    if (av_interleaved_write_frame(oc, &pkt) != 0) {
+        fprintf(stderr, "Error while writing audio frame\n");
+        exit(1);
+    }
+}
+
+static void close_audio(AVFormatContext *oc, AVStream *st)
+{
+    avcodec_close(st->codec);
+
+    av_free(samples);
+    av_free(audio_outbuf);
+}
+
+/**************************************************************/
+/* video output */
+
+AVFrame *picture, *tmp_picture;
+uint8_t *video_outbuf;
+int frame_count, video_outbuf_size;
+
+/* add a video output stream */
+static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
+{
+    AVCodecContext *c;
+    AVStream *st;
+
+    st = av_new_stream(oc, 0);
+    if (!st) {
+        fprintf(stderr, "Could not alloc stream\n");
+        exit(1);
+    }
+
+    c = st->codec;
+    c->codec_id = codec_id;
+    c->codec_type = AVMEDIA_TYPE_VIDEO;
+
+    /* 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 */
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
+        exit(1);
+    }
+
+    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);
+    }
+
+    /* 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);
+        }
+    }
+}
+
+/* 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++) {
+            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++) {
+            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
+            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
+        }
+    }
+}
+
+static void write_video_frame(AVFormatContext *oc, AVStream *st)
+{
+    int out_size, ret;
+    AVCodecContext *c;
+    static struct SwsContext *img_convert_ctx;
+
+    c = st->codec;
+
+    if (frame_count >= STREAM_NB_FRAMES) {
+        /* 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 != PIX_FMT_YUV420P) {
+            /* as we only generate a YUV420P picture, we must convert it
+               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(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(picture, frame_count, c->width, c->height);
+        }
+    }
+
+
+    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+        /* raw video case. The API will change slightly in the near
+           futur for that */
+        AVPacket pkt;
+        av_init_packet(&pkt);
+
+        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 */
+        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);
+
+            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 in the media file */
+            ret = av_interleaved_write_frame(oc, &pkt);
+        } else {
+            ret = 0;
+        }
+    }
+    if (ret != 0) {
+        fprintf(stderr, "Error while writing video frame\n");
+        exit(1);
+    }
+    frame_count++;
+}
+
+static void close_video(AVFormatContext *oc, AVStream *st)
+{
+    avcodec_close(st->codec);
+    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);
+}
+
+/**************************************************************/
+/* media file output */
+
+int main(int argc, char **argv)
+{
+    const char *filename;
+    AVOutputFormat *fmt;
+    AVFormatContext *oc;
+    AVStream *audio_st, *video_st;
+    double audio_pts, video_pts;
+    int i;
+
+    /* 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"
+               "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"
+               "\n", argv[0]);
+        exit(1);
+    }
+
+    filename = argv[1];
+
+    /* allocate the output media context */
+    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
+    if (!oc) {
+        printf("Could not deduce output format from file extension: using MPEG.\n");
+        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
+    }
+    if (!oc) {
+        exit(1);
+    }
+    fmt= oc->oformat;
+
+    /* 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 != CODEC_ID_NONE) {
+        video_st = add_video_stream(oc, fmt->video_codec);
+    }
+    if (fmt->audio_codec != CODEC_ID_NONE) {
+        audio_st = add_audio_stream(oc, fmt->audio_codec);
+    }
+
+    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)) {
+        if (avio_open(&oc->pb, filename, AVIO_WRONLY) < 0) {
+            fprintf(stderr, "Could not open '%s'\n", filename);
+            exit(1);
+        }
+    }
+
+    /* write the stream header, if any */
+    av_write_header(oc);
+
+    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;
+        else
+            video_pts = 0.0;
+
+        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
+            (!video_st || video_pts >= STREAM_DURATION))
+            break;
+
+        /* write interleaved audio and video frames */
+        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
+            write_audio_frame(oc, audio_st);
+        } else {
+            write_video_frame(oc, video_st);
+        }
+    }
+
+    /* 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 */
+    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++) {
+        av_freep(&oc->streams[i]->codec);
+        av_freep(&oc->streams[i]);
+    }
+
+    if (!(fmt->flags & AVFMT_NOFILE)) {
+        /* close the output file */
+        avio_close(oc->pb);
+    }
+
+    /* free the stream */
+    av_free(oc);
+
+    return 0;
+}
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -1,521 +0,0 @@
-/*
- * Copyright (c) 2003 Fabrice Bellard
- *
- * 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 API example.
- *
- * Output a media file in any supported libavformat format.
- * The default codecs are used.
- * @example doc/examples/muxing.c
- */
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <math.h>
-
-#include <libavutil/mathematics.h>
-#include <libavformat/avformat.h>
-#include <libswscale/swscale.h>
-
-/* 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 */
-
-static int sws_flags = SWS_BICUBIC;
-
-/**************************************************************/
-/* audio output */
-
-static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
-
-/* Add an output stream. */
-static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
-                            enum AVCodecID 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);
-    if (!st) {
-        fprintf(stderr, "Could not allocate stream\n");
-        exit(1);
-    }
-    st->id = oc->nb_streams-1;
-    c = st->codec;
-
-    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;
-
-    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. */
-    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)
-{
-    AVCodecContext *c;
-    int ret;
-
-    c = st->codec;
-
-    /* open it */
-    ret = avcodec_open2(c, codec, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
-        exit(1);
-    }
-
-    /* init signal generator */
-    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_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);
-    }
-}
-
-/* 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;
-    int16_t *q;
-
-    q = samples;
-    for (j = 0; j < frame_size; j++) {
-        v = (int)(sin(t) * 10000);
-        for (i = 0; i < nb_channels; i++)
-            *q++ = v;
-        t     += tincr;
-        tincr += tincr2;
-    }
-}
-
-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;
-
-    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.stream_index = st->index;
-
-    /* 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));
-        exit(1);
-    }
-    avcodec_free_frame(&frame);
-}
-
-static void close_audio(AVFormatContext *oc, AVStream *st)
-{
-    avcodec_close(st->codec);
-
-    av_free(samples);
-}
-
-/**************************************************************/
-/* video output */
-
-static AVFrame *frame;
-static AVPicture src_picture, dst_picture;
-static int frame_count;
-
-static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
-{
-    int ret;
-    AVCodecContext *c = st->codec;
-
-    /* open the codec */
-    ret = avcodec_open2(c, codec, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
-        exit(1);
-    }
-
-    /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\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));
-            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)
-{
-    int x, y, i;
-
-    i = frame_index;
-
-    /* Y */
-    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++) {
-            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
-            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
-        }
-    }
-}
-
-static void write_video_frame(AVFormatContext *oc, AVStream *st)
-{
-    int ret;
-    static struct SwsContext *sws_ctx;
-    AVCodecContext *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. */
-    } else {
-        if (c->pix_fmt != AV_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");
-                    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);
-        } else {
-            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
-        }
-    }
-
-    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* Raw video case - directly store the picture in the packet */
-        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);
-
-        ret = av_interleaved_write_frame(oc, &pkt);
-    } else {
-        /* encode the image */
-        AVPacket pkt;
-        int got_output;
-
-        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)
-                pkt.flags |= AV_PKT_FLAG_KEY;
-
-            pkt.stream_index = st->index;
-
-            /* Write the compressed frame to 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));
-        exit(1);
-    }
-    frame_count++;
-}
-
-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);
-}
-
-/**************************************************************/
-/* media file output */
-
-int main(int argc, char **argv)
-{
-    const char *filename;
-    AVOutputFormat *fmt;
-    AVFormatContext *oc;
-    AVStream *audio_st, *video_st;
-    AVCodec *audio_codec, *video_codec;
-    double audio_pts, video_pts;
-    int ret, i;
-
-    /* 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"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    filename = argv[1];
-
-    /* allocate the output media context */
-    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
-    if (!oc) {
-        printf("Could not deduce output format from file extension: using MPEG.\n");
-        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
-    }
-    if (!oc) {
-        return 1;
-    }
-    fmt = oc->oformat;
-
-    /* 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->audio_codec != AV_CODEC_ID_NONE) {
-        audio_st = add_stream(oc, &audio_codec, 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);
-
-    /* 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));
-            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. */
-        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;
-        else
-            video_pts = 0.0;
-
-        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
-            (!video_st || video_pts >= STREAM_DURATION))
-            break;
-
-        /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
-            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);
-        }
-    }
-
-    /* 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(). */
-    av_write_trailer(oc);
-
-    /* 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++) {
-        av_freep(&oc->streams[i]->codec);
-        av_freep(&oc->streams[i]);
-    }
-
-    if (!(fmt->flags & AVFMT_NOFILE))
-        /* Close the output file. */
-        avio_close(oc->pb);
-
-    /* free the stream */
-    av_free(oc);
-
-    return 0;
-}
--- a/doc/examples/resampling_audio.c
+++ b/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;
-}
--- a/doc/examples/scaling_video.c
+++ b/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;
-}
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -11,6 +11,22 @@

@chapter General Questions

+@section When will the next FFmpeg version be released? / Why are FFmpeg releases so few and far between?
+
+Like most open source projects FFmpeg suffers from a certain lack of
+manpower. For this reason the developers have to prioritize the work
+they do and putting out releases is not at the top of the list, fixing
+bugs and reviewing patches takes precedence. Please don't complain or
+request more timely and/or frequent releases unless you are willing to
+help out creating them.
+
+@section I have a problem with an old version of FFmpeg; where should I report it?
+Nowhere. We do not support old FFmpeg versions in any way, we simply lack
+the time, motivation and manpower to do so. If you have a problem with an
+old version of FFmpeg, upgrade to the latest git snapshot. If you
+still experience the problem, then you can report it according to the
+guidelines in @url{http://ffmpeg.org/bugreports.html}.
+
@section Why doesn't FFmpeg support feature [xyz]?

 Because no one has taken on that task yet. FFmpeg development is
@@ -24,6 +40,30 @@ No. Windows DLLs are not portable, bloated and often slow.
 Moreover FFmpeg strives to support all codecs natively.
 A DLL loader is not conducive to that goal.

+@section My bug report/mail to ffmpeg-devel/user has not received any replies.
+
+Likely reasons
+@itemize
+@item We are busy and haven't had time yet to read your report or
+investigate the issue.
+@item You didn't follow @url{http://ffmpeg.org/bugreports.html}.
+@item You didn't use git HEAD.
+@item You reported a segmentation fault without gdb output.
+@item You describe a problem but not how to reproduce it.
+@item It's unclear if you use ffmpeg as command line tool or use
+libav* from another application.
+@item You speak about a video having problems on playback but
+not what you use to play it.
+@item We have no faint clue what you are talking about besides
+that it is related to FFmpeg.
+@end itemize
+
+@section Is there a forum for FFmpeg? I do not like mailing lists.
+
+You may view our mailing lists with a more forum-alike look here:
+@url{http://dir.gmane.org/gmane.comp.video.ffmpeg.user},
+but, if you post, please remember that our mailing list rules still apply there.
+
@section I cannot read this file although this format seems to be supported by ffmpeg.

 Even if ffmpeg can read the container format, it may not support all its
@@ -79,23 +119,11 @@ 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?

-Try a @code{make distclean} in the ffmpeg source directory before the build.
-If this does not help see
+Try a @code{make distclean} in the ffmpeg source directory before the build. If this does not help see
 (@url{http://ffmpeg.org/bugreports.html}).

@section How do I encode single pictures into movies?
@@ -110,16 +138,7 @@ 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 +161,6 @@ 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:
@@ -161,15 +174,15 @@ The @file{movie.mpg} used as input will be converted to

 Instead of relying on file format self-recognition, you may also use
@table @option
-@item -c:v ppm
-@item -c:v png
-@item -c:v mjpeg
+@item -vcodec ppm
+@item -vcodec png
+@item -vcodec mjpeg
@end table
 to force the encoding.

 Applying that to the previous example:
@example
-  ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+  ffmpeg -i movie.mpg -f image2 -vcodec mjpeg menu%d.jpg
@end example

 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -188,21 +201,59 @@ Use @file{-} as file name.

 Try '-f image2 test%d.jpg'.

-@section Why can I not change the frame rate?
+@section Why can I not change the framerate?

-Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
-Choose a different codec with the -c:v command line option.
+Some codecs, like MPEG-1/2, only allow a small number of fixed framerates.
+Choose a different codec with the -vcodec command line option.

@section How do I encode Xvid or DivX video with ffmpeg?

 Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
 standard (note that there are many other coding formats that use this
-same standard). Thus, use '-c:v mpeg4' to encode in these formats. The
+same standard). Thus, use '-vcodec mpeg4' to encode in these formats. The
 default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
 a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
 force the fourcc 'xvid' to be stored as the video fourcc rather than the
 default.

+@section How do I encode videos which play on the iPod?
+
+@table @option
+@item needed stuff
+-acodec libfaac -vcodec mpeg4 width<=320 height<=240
+@item working stuff
+mv4, title
+@item non-working stuff
+B-frames
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128k -vcodec mpeg4 -b 1200k -mbd 2 -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X output.mp4
+@end table
+
+@section How do I encode videos which play on the PSP?
+
+@table @option
+@item needed stuff
+-acodec libfaac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp
+@item working stuff
+mv4, title
+@item non-working stuff
+B-frames
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128k -vcodec mpeg4 -b 1200k -ar 24000 -mbd 2 -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp output.mp4
+@item needed stuff for H.264
+-acodec libfaac -vcodec libx264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
+@item working stuff for H.264
+title, loop filter
+@item non-working stuff for H.264
+CAVLC
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128k -vcodec libx264 -b 1200k -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
+@item higher resolution for newer PSP firmwares, width<=480, height<=272
+-vcodec libx264 -level 21 -coder 1 -f psp
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128k -ac 2 -ar 48000 -vcodec libx264 -level 21 -b 640k -coder 1 -f psp -flags +loop -trellis 2 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -s 480x272 output.mp4
+@end table
+
@section Which are good parameters for encoding high quality MPEG-4?

 '-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
@@ -234,68 +285,12 @@ Just create an "input.avs" text file with this single line ...
  ffmpeg -i input.avs
@end example

-For ANY other help on Avisynth, please visit the
-@uref{http://www.avisynth.org/, Avisynth homepage}.
+For ANY other help on Avisynth, please visit @url{http://www.avisynth.org/}.

@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 +298,28 @@ 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 -sameq intermediate1.mpg
+ffmpeg -i input2.avi -sameq 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 -sameq 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{-sameq} 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 -sameq -y intermediate1.mpg < /dev/null &
+ffmpeg -i input2.avi -sameq -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
+ffmpeg -f mpeg -i - -sameq -vcodec 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 +327,7 @@ 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
@@ -360,90 +344,33 @@ 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
+       -sameq -y output.flv
 rm temp[12].[av] all.[av]
@end example

-@section -profile option fails when encoding H.264 video with AAC audio
+@section The ffmpeg program does not respect the -maxrate setting, some frames are bigger than maxrate/fps.

-@command{ffmpeg} prints an error like
+Read the MPEG spec about video buffer verifier.

-@example
-Undefined constant or missing '(' in 'baseline'
-Unable to parse option value "baseline"
-Error setting option profile to value baseline.
-@end example
+@section I want CBR, but no matter what I do frame sizes differ.

-Short answer: write @option{-profile:v} instead of @option{-profile}.
+You do not understand what CBR is, please read the MPEG spec.
+Read about video buffer verifier and constant bitrate.
+The one sentence summary is that there is a buffer and the input rate is
+constant, the output can vary as needed.

-Long answer: this happens because the @option{-profile} option can apply to both
-video and audio.  Specifically the AAC encoder also defines some profiles, none
-of which are named @var{baseline}.
+@section How do I check if a stream is CBR?

-The solution is to apply the @option{-profile} option to the video stream only
-by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
-Appending @code{:v} to it will do exactly that.
+To quote the MPEG-2 spec:
+"There is no way to tell that a bitstream is constant bitrate without
+examining all of the vbv_delay values and making complicated computations."

-@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
-
-Use @option{-dumpgraph -} to find out exactly where the channel layout is
-lost.
-
-Most likely, it is through @code{auto-inserted aconvert}. Try to understand
-why the converting filter was needed at that place.
-
-Just before the output is a likely place, as @option{-f lavfi} currently
-only support packed S16.
-
-Then insert the correct @code{aconvert} explicitly in the filter graph,
-specifying the exact format.
-
-@example
-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}).

@@ -455,19 +382,42 @@ 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?

 No. These tools are too bloated and they complicate the build.

-@section Why not rewrite FFmpeg in object-oriented C++?
+@section Why not rewrite ffmpeg in object-oriented C++?

 FFmpeg is already organized in a highly modular manner and does not need to
 be rewritten in a formal object language. Further, many of the developers
 favor straight C; it works for them. For more arguments on this matter,
-read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
+read "Programming Religion" at (@url{http://www.tux.org/lkml/#s15}).

@section Why are the ffmpeg programs devoid of debugging symbols?

@@ -478,28 +428,18 @@ you need the debug information, use the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?

 Yes, as long as the code is optional and can easily and cleanly be placed
-under #if CONFIG_GPL without breaking anything. So, for example, a new codec
+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.
+@section I want to compile xyz.c alone but my compiler produced many errors.

-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.
+Common code is in its own files in libav* and is used by the individual
+codecs. They will not work without the common parts, you have to compile
+the whole libav*. If you wish, disable some parts with configure switches.
+You can also try to hack it and remove more, but if you had problems fixing
+the compilation failure then you are probably not qualified for this.

-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.
+@section I'm using libavcodec from within my C++ application but the linker complains about missing symbols which seem to be available.

 FFmpeg is a pure C project, so to use the libraries within your C++ application
 you need to explicitly state that you are using a C library. You can do this by
@@ -507,15 +447,18 @@ encompassing your FFmpeg includes using @code{extern "C"}.

 See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}

-@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
-
-FFmpeg is a pure C project using C99 math features, in order to enable C++
-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 I get "No compatible shell script interpreter found." in MSys.
+
+The standard MSys bash (2.04) is broken. You need to install 2.05 or later.
+
+@section I get "./configure: line <xxx>: pr: command not found" in MSys.
+
+The standard MSys install doesn't come with pr. You need to get it from the coreutils package.

@section Where can I find libav* headers for Pascal/Delphi?

@@ -531,24 +474,12 @@ Even if peculiar since it is network oriented, RTP is a container like any
 other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
 In this specific case please look at RFC 4629 to see how it should be done.

-@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
+@section AVStream.r_frame_rate is wrong, it is much larger than the framerate.

-r_frame_rate is NOT the average frame rate, it is the smallest frame rate
+r_frame_rate is NOT the average framerate, it is the smallest framerate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
 For example, if you have mixed 25 and 30 fps content, then r_frame_rate
 will be 150.

-@section Why is @code{make fate} not running all tests?
-
-Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
-or @code{FATE_SAMPLES} environment variable or the @code{--samples}
-@command{configure} option is set to the right path.
-
-@section Why is @code{make fate} not finding the samples?
-
-Do you happen to have a @code{~} character in the samples path to indicate a
-home directory? The value is used in ways where the shell cannot expand it,
-causing FATE to not find files. Just replace @code{~} by the full path.
-
@bye
--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -1,194 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Automated Testing Environment
-@titlepage
-@center @titlefont{FFmpeg Automated Testing Environment}
-@end titlepage
-
-@node Top
-@top
-
-@contents
-
-@chapter Introduction
-
-  FATE is an extended regression suite on the client-side and a means
-for results aggregation and presentation on the server-side.
-
-  The first part of this document explains how you can use FATE from
-your FFmpeg source directory to test your ffmpeg binary. The second
-part describes how you can run FATE to submit the results to FFmpeg's
-FATE server.
-
-  In any way you can have a look at the publicly viewable FATE results
-by visiting this website:
-
-  @url{http://fate.ffmpeg.org/}
-
-  This is especially recommended for all people contributing source
-code to FFmpeg, as it can be seen if some test on some platform broke
-with there recent contribution. This usually happens on the platforms
-the developers could not test on.
-
-  The second part of this document describes how you can run FATE to
-submit your results to FFmpeg's FATE server. If you want to submit your
-results be sure to check that your combination of CPU, OS and compiler
-is not already listed on the above mentioned website.
-
-  In the third part you can find a comprehensive listing of FATE makefile
-targets and variables.
-
-
-@chapter Using FATE from your FFmpeg source directory
-
-  If you want to run FATE on your machine you need to have the samples
-in place. You can get the samples via the build target fate-rsync.
-Use this command from the top-level source directory:
-
-@example
-make fate-rsync SAMPLES=fate-suite/
-make fate       SAMPLES=fate-suite/
-@end example
-
-  The above commands set the samples location by passing a makefile
-variable via command line. It is also possible to set the samples
-location at source configuration time by invoking configure with
-`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the SAMPLES makefile
-variable. This is illustrated by the following commands:
-
-@example
-./configure --samples=fate-suite/
-make fate-rsync
-make fate
-@end example
-
-  Yet another way to tell FATE about the location of the sample
-directory is by making sure the environment variable FATE_SAMPLES
-contains the path to your samples directory. This can be achieved
-by e.g. putting that variable in your shell profile or by setting
-it in your interactive session.
-
-@example
-FATE_SAMPLES=fate-suite/ make fate
-@end example
-
-@float NOTE
-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
-to be invoked with a configuration file as its first argument.
-
-@example
-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}.
-
-@ifhtml
-  The mentioned configuration template is also available here:
-@verbatiminclude fate_config.sh.template
-@end ifhtml
-
-  Create a configuration that suits your needs, based on the configuration
-template. The `slot' configuration variable can be any string that is not
-yet used, but it is suggested that you name it adhering to the following
-pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-itself will be sourced in a shell script, therefore all shell features may
-be used. This enables you to setup the environment as you need it for your
-build.
-
-  For your first test runs the `fate_recv' variable should be empty or
-commented out. This will run everything as normal except that it will omit
-the submission of the results to the server. The following files should be
-present in $workdir as specified in the configuration file:
-
-@itemize
-    @item configure.log
-    @item compile.log
-    @item test.log
-    @item report
-    @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}.
-
-  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
-of the server and to accept its host key. This can usually be achieved by
-running your SSH client manually and killing it after you accepted the key.
-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.
-
-
-@chapter FATE makefile targets and variables
-
-@section Makefile targets
-
-@table @option
-@item fate-rsync
-    Download/synchronize sample files to the configured samples directory.
-
-@item fate-list
-    Will list all fate/regression test targets.
-
-@item fate
-    Run the FATE test suite (requires the fate-suite dataset).
-@end table
-
-@section Makefile variables
-
-@table @option
-@item V
-    Verbosity level, can be set to 0, 1 or 2.
-    @itemize
-        @item 0: show just the test arguments
-        @item 1: show just the command used in the test
-        @item 2: show everything
-    @end itemize
-
-@item SAMPLES
-    Specify or override the path to the FATE samples at make time, it has a
-    meaning only while running the regression tests.
-
-@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
-make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
-@end example
--- a/doc/fate.txt
+++ b/doc/fate.txt
@@ -0,0 +1,45 @@
+FATE Automated Testing Environment
+
+FATE provides a regression testsuite that can be run locally or configured
+to send reports to fate.ffmpeg.org.
+In order to run, it needs a large amount of data (samples and references)
+that is provided separately from the actual source distribution.
+
+Use the following command to get the fate test samples
+
+# make fate-rsync SAMPLES=fate-suite/
+
+To inform the build system about the testsuite location, pass
+`--samples=<path to the samples>` to configure or set the SAMPLES Make
+variable or the FATE_SAMPLES environment variable to a suitable value.
+
+For information on how to set up FATE to send results to the official FFmpeg
+testing framework, please refer to the following wiki page:
+http://wiki.multimedia.cx/index.php?title=FATE
+
+FATE Makefile targets:
+
+fate-list
+    Will list all fate/regression test targets.
+
+fate
+    Run the FATE test suite (requires the fate-suite dataset).
+
+Fate Makefile variables:
+
+V
+    Verbosity level, can be set to 0, 1 or 2.
+    * 0: show just the test arguments
+    * 1: show just the command used in the test
+    * 2: show everything
+
+SAMPLES
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.
+
+THREADS
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
+
+Example:
+    make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
--- a/doc/fate_config.sh.template
+++ b/doc/fate_config.sh.template
@@ -1,25 +0,0 @@
-slot=                                    # some unique identifier
-repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
-samples=                                 # path to samples directory
-workdir=                                 # directory in which to do all the work
-#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
-comment=                                 # optional description
-
-# the following are optional and map to configure options
-arch=
-cpu=
-cross_prefix=
-cc=
-target_os=
-sysroot=
-target_exec=
-target_path=
-extra_cflags=
-extra_ldflags=
-extra_libs=
-extra_conf=     # extra configure options not covered above
-
-#make=          # name of GNU make if not 'make'
-makeopts=       # extra options passed to 'make'
-#tar=           # command to create a tar archive from its arguments on stdout,
-                # defaults to 'tar c'
--- a/doc/ffmpeg-bitstream-filters.texi
+++ b/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
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
--- a/doc/ffmpeg-devices.texi
+++ b/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
--- a/doc/ffmpeg-filters.texi
+++ b/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
--- a/doc/ffmpeg-formats.texi
+++ b/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
--- a/doc/ffmpeg-mt-authorship.txt
+++ b/doc/ffmpeg-mt-authorship.txt
--- a/doc/ffmpeg-protocols.texi
+++ b/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
--- a/doc/ffmpeg-resampler.texi
+++ b/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
--- a/doc/ffmpeg-scaler.texi
+++ b/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
--- a/doc/ffmpeg-utils.texi
+++ b/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
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
--- a/doc/ffmpeg.txt
+++ b/doc/ffmpeg.txt
@@ -1,47 +0,0 @@
-                                                                                                                                       :
-                                             ffmpeg.c                                                                                  :       libav*
-                                             ========                                                                                  :       ======
-                                                                                                                                       :
-                                                                                                                                       :
-                                                                                                       --------------------------------:---> AVStream...
-                                                                    InputStream input_streams[]      /                                 :
-                                                                                                    /                                  :
-                    InputFile input_files[]                         +==========================+   /   ^                               :
-                                                          ------> 0 |      : st ---:-----------:--/    :                               :
-                 ^  +------+-----------+-----+          /           +--------------------------+       :                               :
-                 :  |      :ist_index--:-----:---------/          1 |      : st    :           |       :                               :
-                 :  +------+-----------+-----+                      +==========================+       :                               :
- nb_input_files  :  |      :ist_index--:-----:------------------> 2 |      : st    :           |       :                               :
-                 :  +------+-----------+-----+                      +--------------------------+       :  nb_input_streams             :
-                 :  |      :ist_index  :     |                    3 |            ...           |       :                               :
-                 v  +------+-----------+-----+                      +--------------------------+       :                               :
-                                                              --> 4 |                          |       :                               :
-                                                             |      +--------------------------+       :                               :
-                                                             |    5 |                          |       :                               :
-                                                             |      +==========================+       v                               :
-                                                             |                                                                         :
-                                                             |                                                                         :
-                                                             |                                                                         :
-                                                             |                                                                         :
-                                                              ---------                                --------------------------------:---> AVStream...
-                                                                        \                            /                                 :
-                                                                    OutputStream output_streams[]   /                                  :
-                                                                          \                        /                                   :
-                                                                    +======\======================/======+      ^                      :
-                                                          ------> 0 |   : source_index  : st-:---        |      :                      :
-                    OutputFile output_files[]           /           +------------------------------------+      :                      :
-                                                       /          1 |   :               :    :           |      :                      :
-                 ^  +------+------------+-----+       /             +------------------------------------+      :                      :
-                 :  |      : ost_index -:-----:------/            2 |   :               :    :           |      :                      :
- nb_output_files :  +------+------------+-----+                     +====================================+      :                      :
-                 :  |      : ost_index -:-----|-----------------> 3 |   :               :    :           |      :                      :
-                 :  +------+------------+-----+                     +------------------------------------+      : nb_output_streams    :
-                 :  |      :            :     |                   4 |                                    |      :                      :
-                 :  +------+------------+-----+                     +------------------------------------+      :                      :
-                 :  |      :            :     |                   5 |                                    |      :                      :
-                 v  +------+------------+-----+                     +------------------------------------+      :                      :
-                                                                  6 |                                    |      :                      :
-                                                                    +------------------------------------+      :                      :
-                                                                  7 |                                    |      :                      :
-                                                                    +====================================+      v                      :
-                                                                                                                                       :
--- a/doc/ffplay.texi
+++ b/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
@@ -24,7 +28,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS

-@include avtools-common-opts.texi
+@include fftools-common-opts.texi

@section Main options

@@ -34,9 +38,8 @@ Force displayed width.
@item -y @var{height}
 Force displayed height.
@item -s @var{size}
-Set frame size (WxH or abbreviation), needed for videos which do
-not contain a header with the frame size like raw YUV.  This option
-has been deprecated in favor of private options, try -video_size.
+Set frame size (WxH or abbreviation), needed for videos which don't
+contain a header with the frame size like raw YUV.
@item -an
 Disable audio.
@item -vn
@@ -87,7 +90,6 @@ Read @var{input_file}.
@table @option
@item -pix_fmt @var{format}
 Set pixel format.
-This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
 Show the stream duration, the codec parameters, the current position in
 the stream and the audio/video synchronisation drift.
@@ -130,20 +132,6 @@ 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.
@end table

@section While playing
@@ -176,9 +164,6 @@ Seek backward/forward 10 seconds.
@item down/up
 Seek backward/forward 1 minute.

-@item page down/page up
-Seek backward/forward 10 minutes.
-
@item mouse click
 Seek to percentage in file corresponding to fraction of width.

@@ -186,35 +171,28 @@ 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
--- a/doc/ffprobe.texi
+++ b/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
@@ -36,22 +42,25 @@ for specifying which information to display, and for setting how
 ffprobe will show it.

 ffprobe output is designed to be easily parsable by a textual filter,
-and consists of one or more sections of 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}.
+and consists of one or more sections of the form:
+@example
+[SECTION]
+key1=val1
+...
+keyN=valN
+[/SECTION]
+@end example

 Metadata tags stored in the container or in the streams are recognized
-and printed in the corresponding "FORMAT" or "STREAM" section.
+and printed in the corresponding "FORMAT" or "STREAM" section, and
+are prefixed by the string "TAG:".

@c man end

@chapter Options
@c man begin OPTIONS

-@include avtools-common-opts.texi
+@include fftools-common-opts.texi

@section Main options

@@ -78,51 +87,6 @@ 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}]
-Set the output printing format.
-
-@var{writer_name} specifies the name of the writer, and
-@var{writer_options} specifies the options to be passed to the writer.
-
-For example for printing the output in JSON format, specify:
-@example
-print_format json
-@end example
-
-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.
-
-The error information is printed within a section with name "ERROR".
-
@item -show_format
 Show information about the container format of the input multimedia
 stream.
@@ -130,64 +94,6 @@ 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.
@@ -195,13 +101,6 @@ stream.
 The information for each single packet is printed within a dedicated
 section with name "PACKET".

-@item -show_frames
-Show information about each frame contained in the input multimedia
-stream.
-
-The information for each single frame is printed within a dedicated
-section with name "FRAME".
-
@item -show_streams
 Show information about each media stream contained in the input
 multimedia stream.
@@ -209,313 +108,30 @@ 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.
-This option is enabled by default, but you may need to disable it
-for specific uses, for example when creating XSD-compliant XML output.
-
-@item -show_program_version
-Show information related to program version.
-
-Version information is printed within a section with name
-"PROGRAM_VERSION".
-
-@item -show_library_versions
-Show information related to library versions.
-
-Version information for each library is printed within a section with
-name "LIBRARY_VERSION".
-
-@item -show_versions
-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}.

@end table
@c man end

-@chapter Writers
-@c man begin WRITERS
-
-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 description of the currently available writers follows.
-
-@section default
-Default format.
-
-Print each section in the form:
-@example
-[SECTION]
-key1=val1
-...
-keyN=valN
-[/SECTION]
-@end example
-
-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.
-
-Each section is printed on a single line.
-If no option is specifid, the output has the form:
-@example
-section|key1=val1| ... |keyN=valN
-@end example
-
-Metadata tags are printed in the corresponding "format" or "stream"
-section. A metadata tag key, if printed, is prefixed by the string
-"tag:".
-
-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).
-
-@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).
-
-@item escape, e
-Set the escape mode to use, default to "c" ("csv" for the @code{csv}
-writer).
-
-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
-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}".
-
-@item csv
-Perform CSV-like escaping, as described in RFC4180.  Strings
-containing a newline ('\n'), a carriage return ('\r'), a double quote
-('"'), or @var{SEP} are enclosed in double-quotes.
-
-@item none
-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.
-
-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
-
-@section json
-JSON based format.
-
-Each section is printed using JSON notation.
-
-The description of the accepted options follows.
-
-@table @option
-
-@item compact, c
-If set to 1 enable compact output, that is each section will be
-printed on a single line. Default value is 0.
-@end table
-
-For more information about JSON, see @url{http://www.json.org/}.
-
-@section xml
-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.
-
-The description of the accepted options follows.
-
-@table @option
-
-@item fully_qualified, q
-If set to 1 specify if the output should be fully qualified. Default
-value is 0.
-This is required for generating an XML file which can be validated
-through an XSD file.
-
-@item xsd_compliant, x
-If set to 1 perform more checks for ensuring that the output is XSD
-compliant. Default value is 0.
-This option automatically sets @option{fully_qualified} to 1.
-@end table
-
-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
-stream details (@option{-show_streams}, see @var{timecode}).
-
-@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
-(@option{-show_format}, see @var{TAG:timecode}).
-
-@end itemize
-@c man end TIMECODE
-
-@chapter See Also
-
-@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
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -1,199 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-
-<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
-    targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
-    xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
-
-    <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
-
-    <xsd:complexType name="ffprobeType">
-        <xsd:sequence>
-            <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
-            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetsType">
-        <xsd:sequence>
-            <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="framesType">
-        <xsd:sequence>
-            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetType">
-      <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
-      <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
-      <xsd:attribute name="pts"           type="xsd:long"  />
-      <xsd:attribute name="pts_time"      type="xsd:float" />
-      <xsd:attribute name="dts"           type="xsd:long"  />
-      <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">
-      <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
-      <xsd:attribute name="pts"           type="xsd:long" />
-      <xsd:attribute name="pts_time"      type="xsd:float"/>
-      <xsd:attribute name="pkt_pts"       type="xsd:long" />
-      <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"  />
-      <xsd:attribute name="height"                 type="xsd:long"  />
-      <xsd:attribute name="pix_fmt"                type="xsd:string"/>
-      <xsd:attribute name="sample_aspect_ratio"    type="xsd:string"/>
-      <xsd:attribute name="pict_type"              type="xsd:string"/>
-      <xsd:attribute name="coded_picture_number"   type="xsd:long"  />
-      <xsd:attribute name="display_picture_number" type="xsd:long"  />
-      <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
-      <xsd:attribute name="top_field_first"        type="xsd:int"   />
-      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
-      <xsd:attribute name="reference"              type="xsd:int"   />
-    </xsd:complexType>
-
-    <xsd:complexType name="streamsType">
-        <xsd:sequence>
-            <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
-        </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"/>
-      <xsd:attribute name="height"               type="xsd:int"/>
-      <xsd:attribute name="has_b_frames"         type="xsd:int"/>
-      <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
-      <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
-      <xsd:attribute name="pix_fmt"              type="xsd:string"/>
-      <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="timecode"             type="xsd:string"/>
-
-      <!-- audio attributes -->
-      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
-      <xsd:attribute name="sample_rate"      type="xsd:int"/>
-      <xsd:attribute name="channels"         type="xsd:int"/>
-      <xsd:attribute name="bits_per_sample"  type="xsd:int"/>
-
-      <xsd:attribute name="id"               type="xsd:string"/>
-      <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">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
-      <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="start_time"       type="xsd:float"/>
-      <xsd:attribute name="duration"         type="xsd:float"/>
-      <xsd:attribute name="size"             type="xsd:long"/>
-      <xsd:attribute name="bit_rate"         type="xsd:long"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="tagType">
-      <xsd:attribute name="key"   type="xsd:string" use="required"/>
-      <xsd:attribute name="value" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="errorType">
-      <xsd:attribute name="code"   type="xsd:int"    use="required"/>
-      <xsd:attribute name="string" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="programVersionType">
-      <xsd:attribute name="version"          type="xsd:string" use="required"/>
-      <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
-      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
-      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="libraryVersionType">
-      <xsd:attribute name="name"        type="xsd:string" use="required"/>
-      <xsd:attribute name="major"       type="xsd:int"    use="required"/>
-      <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">
-        <xsd:sequence>
-          <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-</xsd:schema>
--- a/doc/ffserver.conf
+++ b/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>
+
+
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -9,22 +9,32 @@

@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

 ffserver is a streaming server for both audio and video. It supports
-
 several live feeds, streaming from files and time shifting on live feeds
 (you can seek to positions in the past on each live feed, provided you
 specify a big enough feed storage in ffserver.conf).

+ffserver runs in daemon mode by default; that is, it puts itself in
+the background and detaches from its TTY, unless it is launched in
+debug mode or a NoDaemon option is specified in the configuration
+file.
+
 This documentation covers only the streaming aspects of ffserver /
 ffmpeg. All questions about parameters for ffmpeg, codec questions,
-etc. are not covered here. Read @file{ffmpeg.html} for more
+etc. are not covered here. Read @file{ffmpeg-doc.html} for more
 information.

@section How does it work?
@@ -81,7 +91,7 @@ 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.]

@@ -100,8 +110,8 @@ As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:

@example
-ffserver -f doc/ffserver.conf &
-ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
+./ffserver -f doc/ffserver.conf &
+./ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
@end example

 At this point you should be able to go to your Windows machine and fire up
@@ -136,7 +146,7 @@ that only captures in stereo and also requires that one channel be flipped.
 If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
 starting ffmpeg.

-@subsection The audio and video lose sync after a while.
+@subsection The audio and video loose sync after a while.

 Yes, they do.

@@ -227,23 +237,10 @@ 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

-@include avtools-common-opts.texi
+@include fftools-common-opts.texi

@section Main options

@@ -256,40 +253,26 @@ 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{ffmpeg/doc/ffserver.conf}
+example and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
@end ignore

@bye
--- a/doc/fftools-common-opts.texi
+++ b/doc/fftools-common-opts.texi
@@ -0,0 +1,93 @@
+All the numerical options, if not specified otherwise, accept in input
+a string representing a number, which may contain one of the
+International System number postfixes, for example 'K', 'M', 'G'.
+If 'i' is appended after the postfix, powers of 2 are used instead of
+powers of 10. The 'B' postfix multiplies the value for 8, and can be
+appended after another postfix or used alone. This allows using for
+example 'KB', 'MiB', 'G' and 'B' as postfix.
+
+Options which do not take arguments are boolean options, and set the
+corresponding value to true. They can be set to false by prefixing
+with "no" the option name, for example using "-nofoo" in the
+commandline will set to false the boolean option with name "foo".
+
+@section Generic options
+
+These options are shared amongst the ff* tools.
+
+@table @option
+
+@item -L
+Show license.
+
+@item -h, -?, -help, --help
+Show help.
+
+@item -version
+Show version.
+
+@item -formats
+Show available formats.
+
+The fields preceding the format names have the following meanings:
+@table @samp
+@item D
+Decoding available
+@item E
+Encoding available
+@end table
+
+@item -codecs
+Show available codecs.
+
+The fields preceding the codec names have the following meanings:
+@table @samp
+@item D
+Decoding available
+@item E
+Encoding available
+@item V/A/S
+Video/audio/subtitle codec
+@item S
+Codec supports slices
+@item D
+Codec supports direct rendering
+@item T
+Codec can handle input truncated at random locations instead of only at frame boundaries
+@end table
+
+@item -bsfs
+Show available bitstream filters.
+
+@item -protocols
+Show available protocols.
+
+@item -filters
+Show available libavfilter filters.
+
+@item -pix_fmts
+Show available pixel formats.
+
+@item -loglevel @var{loglevel}
+Set the logging level used by the library.
+@var{loglevel} is a number or a string containing one of the following values:
+@table @samp
+@item quiet
+@item panic
+@item fatal
+@item error
+@item warning
+@item info
+@item verbose
+@item debug
+@end table
+
+By default the program logs to stderr, if coloring is supported by the
+terminal, colors are used to mark errors and warnings. Log coloring
+can be disabled setting the environment variable
+@env{FFMPEG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
+the environment variable @env{FFMPEG_FORCE_COLOR}.
+The use of the environment variable @env{NO_COLOR} is deprecated and
+will be dropped in a following FFmpeg version.
+
+@end table
--- a/doc/filter_design.txt
+++ b/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.
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/general.texi
+++ b/doc/general.texi
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -1,415 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Using git to develop FFmpeg
-
-@titlepage
-@center @titlefont{Using git to develop FFmpeg}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Introduction
-
-This document aims in giving some quick references on a set of useful git
-commands. You should always use the extensive and detailed documentation
-provided directly by git:
-
-@example
-git --help
-man git
-@end example
-
-shows you the available subcommands,
-
-@example
-git <command> --help
-man git-<command>
-@end example
-
-shows information about the subcommand <command>.
-
-Additional information could be found on the
-@url{http://gitref.org, Git Reference} website
-
-For more information about the Git project, visit the
-
-@url{http://git-scm.com/, Git website}
-
-Consult these resources whenever you have problems, they are quite exhaustive.
-
-What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines to ease the contribution to the project
-
-@chapter Basics Usage
-
-@section Get GIT
-
-You can get git from @url{http://git-scm.com/}
-Most distribution and operating system provide a package for it.
-
-
-@section Cloning the source tree
-
-@example
-git clone git://source.ffmpeg.org/ffmpeg <target>
-@end example
-
-This will put the FFmpeg sources into the directory @var{<target>}.
-
-@example
-git clone git@@source.ffmpeg.org:ffmpeg <target>
-@end example
-
-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
-
-@example
-git pull (--rebase)
-@end example
-
-pulls in the latest changes from the tracked branch. The tracked branch
-can be remote. By default the master branch tracks the branch master in
-the remote origin.
-
-@float IMPORTANT
-@command{--rebase} (see below) is recommended.
-@end float
-
-@section Rebasing your local branches
-
-@example
-git pull --rebase
-@end example
-
-fetches the changes from the main repository and replays your local commits
-over it. This is required to keep all your local changes at the top of
-FFmpeg's master tree. The master tree will reject pushes with merge commits.
-
-
-@section Adding/removing files/directories
-
-@example
-git add [-A] <filename/dirname>
-git rm [-r] <filename/dirname>
-@end example
-
-GIT needs to get notified of all changes you make to your working
-directory that makes files appear or disappear.
-Line moves across files are automatically tracked.
-
-
-@section Showing modifications
-
-@example
-git diff <filename(s)>
-@end example
-
-will show all local modifications in your working directory as unified diff.
-
-
-@section Inspecting the changelog
-
-@example
-git log <filename(s)>
-@end example
-
-You may also use the graphical tools like gitview or gitk or the web
-interface available at http://source.ffmpeg.org/
-
-@section Checking source tree status
-
-@example
-git status
-@end example
-
-detects all the changes you made and lists what actions will be taken in case
-of a commit (additions, modifications, deletions, etc.).
-
-
-@section Committing
-
-@example
-git diff --check
-@end example
-
-to double check your changes before committing them to avoid trouble later
-on. All experienced developers do this on each and every commit, no matter
-how small.
-Every one of them has been saved from looking like a fool by this many times.
-It's very easy for stray debug output or cosmetic modifications to slip in,
-please avoid problems through this extra level of scrutiny.
-
-For cosmetics-only commits you should get (almost) empty output from
-
-@example
-git diff -w -b <filename(s)>
-@end example
-
-Also check the output of
-
-@example
-git status
-@end example
-
-to make sure you don't have untracked files or deletions.
-
-@example
-git add [-i|-p|-A] <filenames/dirnames>
-@end example
-
-Make sure you have told git your name and email address
-
-@example
-git config --global user.name "My Name"
-git config --global user.email my@@email.invalid
-@end example
-
-Use @var{--global} to set the global configuration for all your git checkouts.
-
-Git will select the changes to the files for commit. Optionally you can use
-the interactive or the patch mode to select hunk by hunk what should be
-added to the commit.
-
-
-@example
-git commit
-@end example
-
-Git will commit the selected changes to your current local branch.
-
-You will be prompted for a log message in an editor, which is either
-set in your personal configuration file through
-
-@example
-git config --global core.editor
-@end example
-
-or set by one of the following environment variables:
-@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
-
-Log messages should be concise but descriptive. Explain why you made a change,
-what you did will be obvious from the changes themselves most of the time.
-Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
-levels look at and educate themselves while reading through your code. Don't
-include filenames in log messages, Git provides that information.
-
-Possibly make the commit message have a terse, descriptive first line, an
-empty line and then a full description. The first line will be used to name
-the patch by git format-patch.
-
-@section Preparing a patchset
-
-@example
-git format-patch <commit> [-o directory]
-@end example
-
-will generate a set of patches for each commit between @var{<commit>} and
-current @var{HEAD}. E.g.
-
-@example
-git format-patch origin/master
-@end example
-
-will generate patches for all commits on current branch which are not
-present in upstream.
-A useful shortcut is also
-
-@example
-git format-patch -n
-@end example
-
-which will generate patches from last @var{n} commits.
-By default the patches are created in the current directory.
-
-@section Sending patches for review
-
-@example
-git send-email <commit list|directory>
-@end example
-
-will send the patches created by @command{git format-patch} or directly
-generates them. All the email fields can be configured in the global/local
-configuration or overridden by command line.
-Note that this tool must often be installed separately (e.g. @var{git-email}
-package on Debian-based distros).
-
-
-@section Renaming/moving/copying files or contents of files
-
-Git automatically tracks such changes, making those normal commits.
-
-@example
-mv/cp path/file otherpath/otherfile
-git add [-A] .
-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
-
-@example
-git reset <commit>
-@end example
-
-@command{git reset} will uncommit the changes till @var{<commit>} rewriting
-the current branch history.
-
-@example
-git commit --amend
-@end example
-
-allows to amend the last commit details quickly.
-
-@example
-git rebase -i origin/master
-@end example
-
-will replay local commits over the main repository allowing to edit, merge
-or remove some of them in the process.
-
-@float NOTE
-@command{git reset}, @command{git commit --amend} and @command{git rebase}
-rewrite history, so you should use them ONLY on your local or topic branches.
-The main repository will reject those changes.
-@end float
-
-@example
-git revert <commit>
-@end example
-
-@command{git revert} will generate a revert commit. This will not make the
-faulty commit disappear from the history.
-
-@section Pushing changes to remote trees
-
-@example
-git push
-@end example
-
-Will push the changes to the default remote (@var{origin}).
-Git will prevent you from pushing changes if the local and remote trees are
-out of sync. Refer to and to sync the local tree.
-
-@example
-git remote add <name> <url>
-@end example
-
-Will add additional remote with a name reference, it is useful if you want
-to push your local branch for review on a remote host.
-
-@example
-git push <remote> <refspec>
-@end example
-
-Will push the changes to the @var{<remote>} repository.
-Omitting @var{<refspec>} makes @command{git push} update all the remote
-branches matching the local ones.
-
-@section Finding a specific svn revision
-
-Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
-based on a regular expression. see man gitrevisions
-
-@example
-git show :/'as revision 23456'
-@end example
-
-will show the svn changeset @var{r23456}. With older git versions searching in
-the @command{git log} output is the easiest option (especially if a pager with
-search capabilities is used).
-This commit can be checked out with
-
-@example
-git checkout -b svn_23456 :/'as revision 23456'
-@end example
-
-or for git < 1.7.1 with
-
-@example
-git checkout -b svn_23456 $SHA1
-@end example
-
-where @var{$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
-problems with the GIT server.
--- a/doc/git-howto.txt
+++ b/doc/git-howto.txt
@@ -39,17 +39,16 @@ I. BASICS:

 0. Get GIT:

-  Most distributions have a git package, if not
  You can get git from http://git-scm.com/


 1. Cloning the source tree:

-    git clone git://source.ffmpeg.org/ffmpeg <target>
+    git clone git://git.videolan.org/ffmpeg <target>

  This will put the FFmpeg sources into the directory <target>.

-    git clone git@source.ffmpeg.org:ffmpeg <target>
+    git clone git@git.videolan.org:ffmpeg <target>

  This will put the FFmpeg sources into the directory <target> and let
  you push back your changes to the remote repository.
@@ -98,7 +97,7 @@ I. BASICS:
    git log <filename(s)>

  You may also use the graphical tools like gitview or gitk or the web
-  interface available at http://source.ffmpeg.org
+  interface available at http://git.videolan.org

 6. Checking source tree status:

@@ -206,19 +205,8 @@ I. BASICS:

    git format-patch <commit> [-o directory]

-  will generate a set of patches for each commit between <commit> and
-  current HEAD. E.g.
-
-    git format-patch origin/master
-
-  will generate patches for all commits on current branch which are not
-  present in upstream.
-  A useful shortcut is also
-
-    git format-patch -n
-
-  which will generate patches from last n commits.
-  By default the patches are created in the current directory.
+  will generate a set of patches out of the current branch starting from
+  commit. By default the patches are created in the current directory.

 11. Sending patches for review

@@ -227,8 +215,6 @@ I. BASICS:
  will send the patches created by git format-patch or directly generates
  them. All the email fields can be configured in the global/local
  configuration or overridden by command line.
-  Note that this tool must often be installed separately (e.g. git-email
-  package on Debian-based distros).

 12. Pushing changes to remote trees

--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -42,7 +42,7 @@ specify card number or identifier, device number and subdevice number
 To see the list of cards currently recognized by your system check the
 files @file{/proc/asound/cards} and @file{/proc/asound/devices}.

-For example to capture with @command{ffmpeg} from an ALSA device with
+For example to capture with @file{ffmpeg} from an ALSA device with
 card id 0, you may run the command:
@example
 ffmpeg -f alsa -i hw:0 alsaout.wav
@@ -55,114 +55,6 @@ For more information see:

 BSD video input device.

-@section dshow
-
-Windows DirectShow input device.
-
-DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.
-Currently only audio and video devices are supported.
-
-Multiple devices may be opened as separate inputs, but they may also be
-opened on the same input, which should improve synchronism between them.
-
-The input name should be in the format:
-
-@example
-@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}]
-@end example
-
-where @var{TYPE} can be either @var{audio} or @var{video},
-and @var{NAME} is the device's name.
-
-@subsection Options
-
-If no options are specified, the device's defaults are used.
-If the device does not support the requested options, it will
-fail to open.
-
-@table @option
-
-@item video_size
-Set the video size in the captured video.
-
-@item framerate
-Set the framerate in the captured video.
-
-@item sample_rate
-Set the sample rate (in Hz) of the captured audio.
-
-@item sample_size
-Set the sample size (in bits) of the captured audio.
-
-@item channels
-Set the number of channels in the captured audio.
-
-@item list_devices
-If set to @option{true}, print a list of devices and exit.
-
-@item list_options
-If set to @option{true}, print a list of selected device's options
-and exit.
-
-@item video_device_number
-Set video device number for devices with same name (starts at 0,
-defaults to 0).
-
-@item audio_device_number
-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
-
-@itemize
-
-@item
-Print the list of DirectShow supported devices and exit:
-@example
-$ ffmpeg -list_devices true -f dshow -i dummy
-@end example
-
-@item
-Open video device @var{Camera}:
-@example
-$ ffmpeg -f dshow -i video="Camera"
-@end example
-
-@item
-Open second video device with name @var{Camera}:
-@example
-$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
-@end example
-
-@item
-Open video device @var{Camera} and audio device @var{Microphone}:
-@example
-$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
-@end example
-
-@item
-Print the list of supported options in selected device and exit:
-@example
-$ ffmpeg -list_options true -f dshow -i video="Camera"
-@end example
-
-@end itemize
-
@section dv1394

 Linux DV 1394 input device.
@@ -180,78 +72,18 @@ For more detailed information read the file
 Documentation/fb/framebuffer.txt included in the Linux source tree.

 To record from the framebuffer device @file{/dev/fb0} with
-@command{ffmpeg}:
+@file{ffmpeg}:
@example
 ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
@end example

 You can take a single screenshot image with the command:
@example
-ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
+ffmpeg -f fbdev -vframes 1 -r 1 -i /dev/fb0 screenshot.jpeg
@end example

 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.
@@ -269,15 +101,15 @@ device.
 Once you have created one or more JACK readable clients, you need to
 connect them to one or more JACK writable clients.

-To connect or disconnect JACK clients you can use the @command{jack_connect}
-and @command{jack_disconnect} programs, or do it through a graphical interface,
-for example with @command{qjackctl}.
+To connect or disconnect JACK clients you can use the
+@file{jack_connect} and @file{jack_disconnect} programs, or do it
+through a graphical interface, for example with @file{qjackctl}.

 To list the JACK clients and their properties you can invoke the command
-@command{jack_lsp}.
+@file{jack_lsp}.

 Follows an example which shows how to capture a JACK readable client
-with @command{ffmpeg}.
+with @file{ffmpeg}.
@example
 # Create a JACK writable client with name "ffmpeg".
 $ ffmpeg -f jack -i ffmpeg -y out.wav
@@ -301,171 +133,10 @@ $ jack_connect metro:120_bpm ffmpeg:input_1
 For more information read:
@url{http://jackaudio.org/}

-@section lavfi
-
-Libavfilter input virtual device.
-
-This input device reads data from the open output pads of a libavfilter
-filtergraph.
-
-For each filtergraph open output, the input device will create a
-corresponding stream which is mapped to the generated output. Currently
-only video data is supported. The filtergraph is specified through the
-option @option{graph}.
-
-@subsection Options
-
-@table @option
-
-@item graph
-Specify the filtergraph to use as input. Each video open output must be
-labelled by a unique string of the form "out@var{N}", where @var{N} is a
-number starting from 0 corresponding to the mapped input stream
-generated by the device.
-The first unlabelled output is automatically assigned to the "out0"
-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
-
-@itemize
-@item
-Create a color video stream and play it back with @command{ffplay}:
-@example
-ffplay -f lavfi -graph "color=c=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
-@end example
-
-@item
-Create three different video test filtered sources and play them:
-@example
-ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
-@end example
-
-@item
-Read an audio stream from a file using the amovie source and play it
-back with @command{ffplay}:
-@example
-ffplay -f lavfi "amovie=test.wav"
-@end example
-
-@item
-Read an audio stream and a video stream and play it back with
-@command{ffplay}:
-@example
-ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
-@end example
-
-@end itemize
-
@section libdc1394

 IIDC1394 input device, based on libdc1394 and libraw1394.

-@section openal
-
-The OpenAL input device provides audio capture on all systems with a
-working OpenAL 1.1 implementation.
-
-To enable this input device during configuration, you need OpenAL
-headers and libraries installed on your system, and need to configure
-FFmpeg with @code{--enable-openal}.
-
-OpenAL headers and libraries should be provided as part of your OpenAL
-implementation, or as an additional download (an SDK). Depending on your
-installation you may need to specify additional flags via the
-@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build
-system to locate the OpenAL headers and libraries.
-
-An incomplete list of OpenAL implementations follows:
-
-@table @strong
-@item Creative
-The official Windows implementation, providing hardware acceleration
-with supported devices and software fallback.
-See @url{http://openal.org/}.
-@item OpenAL Soft
-Portable, open source (LGPL) software implementation. Includes
-backends for the most common sound APIs on the Windows, Linux,
-Solaris, and BSD operating systems.
-See @url{http://kcat.strangesoft.net/openal.html}.
-@item Apple
-OpenAL is part of Core Audio, the official Mac OS X Audio interface.
-See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
-@end table
-
-This device allows to capture from an audio input device handled
-through OpenAL.
-
-You need to specify the name of the device to capture in the provided
-filename. If the empty string is provided, the device will
-automatically select the default device. You can get the list of the
-supported devices by using the option @var{list_devices}.
-
-@subsection Options
-
-@table @option
-
-@item channels
-Set the number of channels in the captured audio. Only the values
-@option{1} (monaural) and @option{2} (stereo) are currently supported.
-Defaults to @option{2}.
-
-@item sample_size
-Set the sample size (in bits) of the captured audio. Only the values
-@option{8} and @option{16} are currently supported. Defaults to
-@option{16}.
-
-@item sample_rate
-Set the sample rate (in Hz) of the captured audio.
-Defaults to @option{44.1k}.
-
-@item list_devices
-If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}.
-
-@end table
-
-@subsection Examples
-
-Print the list of OpenAL supported devices and exit:
-@example
-$ ffmpeg -list_devices true -f openal -i dummy out.ogg
-@end example
-
-Capture from the OpenAL device @file{DR-BT101 via PulseAudio}:
-@example
-$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg
-@end example
-
-Capture from the default device (note the empty string '' as filename):
-@example
-$ ffmpeg -f openal -i '' out.ogg
-@end example
-
-Capture from two devices simultaneously, writing to two different files,
-within the same @command{ffmpeg} command:
-@example
-$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
-@end example
-Note: not all OpenAL implementations support multiple simultaneous capture -
-try the latest OpenAL Soft if the above does not work.
-
@section oss

 Open Sound System input device.
@@ -474,7 +145,7 @@ The filename to provide to the input device is the device node
 representing the OSS input device, and is usually set to
@file{/dev/dsp}.

-For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the
+For example to grab from @file{/dev/dsp} using @file{ffmpeg} use the
 command:
@example
 ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
@@ -483,89 +154,6 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
 For more information about OSS see:
@url{http://manuals.opensound.com/usersguide/dsp.html}

-@section pulse
-
-pulseaudio input device.
-
-To enable this input device during configuration you need libpulse-simple
-installed in your system.
-
-The filename to provide to the input device is a source device or the
-string "default"
-
-To list the pulse source devices and their properties you can invoke
-the command @command{pactl list sources}.
-
-@example
-ffmpeg -f pulse -i default /tmp/pulse.wav
-@end example
-
-@subsection @var{server} AVOption
-
-The syntax is:
-@example
-server @var{server name}
-@end example
-
-Connects to a specific server.
-
-@subsection @var{name} AVOption
-
-The syntax is:
-@example
-name @var{application name}
-@end example
-
-Specify the application name pulse will use when showing active clients,
-by default it is the LIBAVFORMAT_IDENT string
-
-@subsection @var{stream_name} AVOption
-
-The syntax is:
-@example
-stream_name @var{stream name}
-@end example
-
-Specify the stream name pulse will use when showing active streams,
-by default it is "record"
-
-@subsection @var{sample_rate} AVOption
-
-The syntax is:
-@example
-sample_rate @var{samplerate}
-@end example
-
-Specify the samplerate in Hz, by default 48kHz is used.
-
-@subsection @var{channels} AVOption
-
-The syntax is:
-@example
-channels @var{N}
-@end example
-
-Specify the channels in use, by default 2 (stereo) is set.
-
-@subsection @var{frame_size} AVOption
-
-The syntax is:
-@example
-frame_size @var{bytes}
-@end example
-
-Specify the number of byte per frame, by default it is set to 1024.
-
-@subsection @var{fragment_size} AVOption
-
-The syntax is:
-@example
-fragment_size @var{bytes}
-@end example
-
-Specify the minimal buffering fragment in pulseaudio, it will affect the
-audio latency. By default it is unset.
-
@section sndio

 sndio input device.
@@ -577,15 +165,15 @@ The filename to provide to the input device is the device node
 representing the sndio input device, and is usually set to
@file{/dev/audio0}.

-For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the
+For example to grab from @file{/dev/audio0} using @file{ffmpeg} use the
 command:
@example
 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
@@ -593,33 +181,35 @@ 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 @file{dov4l} for Video4Linux
+devices and the command @file{v4l-info} 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 autodetect 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.
+Follow some usage examples of the video4linux devices with the ff*
+tools.
@example
-# Grab and show the input of a video4linux2 device.
-ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
+# 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 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
+# Grab and show the input of a video4linux2 device, autoadjust size.
+ffplay -f video4linux2 /dev/video0
+
+# Grab and record the input of a video4linux2 device, autoadjust 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
-"video4linux2".
-
@section vfwcap

 VfW (Video for Windows) capture input device.
@@ -641,7 +231,7 @@ The filename passed as input has the syntax:

@var{hostname}:@var{display_number}.@var{screen_number} specifies the
 X11 display name of the screen to grab from. @var{hostname} can be
-omitted, and defaults to "localhost". The environment variable
+ommitted, and defaults to "localhost". The environment variable
@env{DISPLAY} contains the default display name.

@var{x_offset} and @var{y_offset} specify the offsets of the grabbed
@@ -650,68 +240,15 @@ default to 0.

 Check the X11 documentation (e.g. man X) for more detailed information.

-Use the @command{dpyinfo} program for getting basic information about the
+Use the @file{dpyinfo} program for getting basic information about the
 properties of your X11 display (e.g. grep for "name" or "dimensions").

-For example to grab from @file{:0.0} using @command{ffmpeg}:
+For example to grab from @file{:0.0} using @file{ffmpeg}:
@example
 ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
+
+# Grab at position 10,20.
+ffmpeg -f x11grab -25 -s cif -i :0.0+10,20 out.mpg
@end example

-Grab at position @code{10,20}:
-@example
-ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
-@end example
-
-@subsection Options
-
-@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}.
-
-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
-follows only when the mouse pointer reaches within @var{PIXELS} (greater than
-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
-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}.
-
-@item show_region
-Show grabbed region on screen.
-
-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.
-
-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
-@end example
-
-@item video_size
-Set the video frame size. Default value is @code{vga}.
-@end table
-
@c man end INPUT DEVICES
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -5,42 +5,32 @@ NOTE: This is a draft.

 Overview:
 ---------
-
 FFmpeg uses Trac for tracking issues, new issues and changes to
 existing issues can be done through a web interface.
-
-Issues can be different kinds of things we want to keep track of
-but that do not belong into the source tree itself. This includes
-bug reports, patches, feature requests and license violations. We
-might add more items to this list in the future, so feel free to
-propose a new `type of issue' on the ffmpeg-devel mailing list if
-you feel it is worth tracking.
-
 It is possible to subscribe to individual issues by adding yourself to the
-Cc list or to subscribe to the ffmpeg-trac mailing list which receives
-a mail for every change to every issue.
+nosy list or to subscribe to the ffmpeg-issues mailing list which receives
+a mail for every change to every issue. Replies to such mails will also
+be properly added to the respective issue.
 (the above does all work already after light testing)
-
 The subscription URL for the ffmpeg-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
+
+NOTE: issue = (bug report || patch || feature request)

 Type:
 -----
-bug / defect
+bug
    An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
    prevents it from behaving as intended.

-feature request / enhancement
+feature request
    Request of support for encoding or decoding of a new codec, container
    or variant.
    Request of support for more, less or plain different output or behavior
    where the current implementation cannot be considered wrong.

-license violation
-    ticket to keep track of (L)GPL violations of ffmpeg by others
-
 patch
    A patch as generated by diff which conforms to the patch submission and
    development policy.
@@ -61,8 +51,6 @@ important
    the separation to normal is somewhat fuzzy.
    For feature requests this priority would be used for things many people
    want.
-    Regressions also should be marked as important, regressions are bugs that
-    don't exist in a past revision or another branch.

 normal

@@ -92,17 +80,6 @@ closed
    final state


-Analyzed flag:
--------------
-Bugs which have been analyzed and where it is understood what causes them
-and which exact chain of events triggers them. This analysis should be
-available as a message in the bug report.
-Note, do not change the status to analyzed without also providing a clear
-and understandable analysis.
-This state implicates that the bug either has been reproduced or that
-reproduction is not needed as the bug is already understood.
-
-
 Type/Status/Substatus:
 ----------
 */new/new
@@ -130,6 +107,24 @@ Type/Status/Substatus:
    Issues for which some information has been requested by the developers,
    but which has not been provided by anyone within reasonable time.

+bug/open/reproduced
+    Bugs which have been reproduced.
+
+bug/open/analyzed
+    Bugs which have been analyzed and where it is understood what causes them
+    and which exact chain of events triggers them. This analysis should be
+    available as a message in the bug report.
+    Note, do not change the status to analyzed without also providing a clear
+    and understandable analysis.
+    This state implicates that the bug either has been reproduced or that
+    reproduction is not needed as the bug is already understood.
+
+bug/open/needs_more_info
+    Bug reports which are incomplete and or where more information is needed
+    from the submitter or another person who can provide it.
+    This state implicates that the bug has not been analyzed or reproduced.
+    Note, the idea behind needs_more_info is to offload work from the
+    developers to the users whenever possible.

 bug/closed/fixed
    Bugs which have to the best of our knowledge been fixed.
@@ -163,6 +158,10 @@ patch/closed/applied
 patch/closed/rejected
    Patches which have been rejected.

+feature_request/open/needs_more_info
+    Feature requests where it is not clear what exactly is wanted
+    (these also could be closed as invalid ...).
+
 feature_request/closed/implemented
    Feature requests which have been implemented.

@@ -174,10 +173,12 @@ Note, please do not use type-status-substatus combinations other than the
 above without asking on ffmpeg-dev first!

 Note2, if you provide the requested info do not forget to remove the
-needs_more_info substatus.
+needs_more_info substate.

-Component:
----------
+Topic:
+------
+A topic is a tag you should add to your issue in order to make grouping them
+easier.

 avcodec
    issues in libavcodec/*
@@ -197,9 +198,6 @@ ffmpeg
 ffplay
    issues in or related to ffplay.c

-ffprobe
-    issues in or related to ffprobe.c
-
 ffserver
    issues in or related to ffserver.c

@@ -207,7 +205,7 @@ build system
    issues in or related to configure/Makefile

 regression
-    bugs which were not present in a past revision
+    bugs which were working in a past revision

-trac
+roundup
    issues related to our issue tracker
--- a/doc/libavcodec.texi
+++ b/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
--- a/doc/libavdevice.texi
+++ b/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
--- a/doc/libavfilter.texi
+++ b/doc/libavfilter.texi
@@ -9,36 +9,97 @@

@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
+Integrating libavfilter into the main FFmpeg repository is a work in
+progress. If you wish to try the unfinished development code of
+libavfilter then check it out from the libavfilter repository into
+some directory of your choice by:

-@chapter See Also
+@example
+   svn checkout svn://svn.ffmpeg.org/soc/libavfilter
+@end example

-@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
+And then read the README file in the top directory to learn how to
+integrate it into ffmpeg and ffplay.

-@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
+But note that there may still be serious bugs in the code and its API
+and ABI should not be considered stable yet!

-@include authors.texi
+@chapter Tutorial

-@ignore
+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:

-@setfilename libavfilter
-@settitle multimedia filtering library
+@example
+input --> split --> fifo -----------------------> overlay --> output
+            |                                        ^
+            |                                        |
+            +------> fifo --> crop --> vflip --------+
+@end example

-@end ignore
+splits the stream in two streams, sends one stream through the crop filter
+and the vflip filter before merging it back with the other stream by
+overlaying it on top. You can use the following command to achieve this:
+
+@example
+./ffmpeg -i in.avi -s 240x320 -vf "[in] split [T1], fifo, [T2] overlay= 0:240 [out]; [T1] fifo, crop=0:0:-1:240, vflip [T2]
+@end example
+
+where input_video.avi has a vertical resolution of 480 pixels. The
+result will be that in output the top half of the video is mirrored
+onto the bottom half.
+
+Video filters are loaded using the @var{-vf} option passed to
+ffmpeg or to ffplay. Filters in the same linear chain are separated by
+commas. In our example, @var{split, fifo, overlay} are in one linear
+chain, and @var{fifo, crop, vflip} are in another. The points where
+the linear chains join are labeled by names enclosed in square
+brackets. In our example, that is @var{[T1]} and @var{[T2]}. The magic
+labels @var{[in]} and @var{[out]} are the points where video is input
+and output.
+
+Some filters take in input a list of parameters: they are specified
+after the filter name and an equal sign, and are separated each other
+by a semicolon.
+
+There exist so-called @var{source filters} that do not have a video
+input, and we expect in the future some @var{sink filters} that will
+not have video output.
+
+@chapter graph2dot
+
+The @file{graph2dot} program included in the FFmpeg @file{tools}
+directory can be used to parse a filter graph description and issue a
+corresponding textual representation in the dot language.
+
+Invoke the command:
+@example
+graph2dot -h
+@end example
+
+to see how to use @file{graph2dot}.
+
+You can then pass the dot description to the @file{dot} program (from
+the graphviz suite of programs) and obtain a graphical representation
+of the filter graph.
+
+For example the sequence of commands:
+@example
+echo @var{GRAPH_DESCRIPTION} | \
+tools/graph2dot -o graph.tmp && \
+dot -Tpng graph.tmp -o graph.png && \
+display graph.png
+@end example
+
+can be used to create and display an image representing the graph
+described by the @var{GRAPH_DESCRIPTION} string.
+
+@include filters.texi

@bye
--- a/doc/libavformat.texi
+++ b/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
--- a/doc/libavutil.texi
+++ b/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
--- a/doc/libswresample.texi
+++ b/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
--- a/doc/libswscale.texi
+++ b/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
--- a/doc/mips.txt
+++ b/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
--- a/doc/muxers.texi
+++ b/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

@@ -60,152 +43,53 @@ You can print the CRC to stdout with the command:
 ffmpeg -i INPUT -f crc -
@end example

-You can select the output format of each frame with @command{ffmpeg} by
+You can select the output format of each frame with @file{ffmpeg} by
 specifying the audio and video codec and format. For example to
 compute the CRC of the input audio converted to PCM unsigned 8-bit
 and the input video converted to MPEG-2 video, use the command:
@example
-ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
+ffmpeg -i INPUT -acodec pcm_u8 -vcodec mpeg2video -f crc -
@end example

-See also the @ref{framecrc} muxer.
+See also the @code{framecrc} muxer (@pxref{framecrc}).

@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 @file{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:
@example
-ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
+ffmpeg -i INPUT -acodec pcm_u8 -vcodec mpeg2video -f framecrc -
@end example

-See also the @ref{crc} muxer.
+See also the @code{crc} muxer (@pxref{crc}).

-@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

 Image file muxer.
@@ -235,142 +119,34 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
 form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
 etc.

-The following example shows how to use @command{ffmpeg} for creating a
+The following example shows how to use @file{ffmpeg} for creating a
 sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
 taking one image every second from the input video:
@example
-ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg'
+ffmpeg -i in.avi -r 1 -f image2 'img-%03d.jpeg'
@end example

-Note that with @command{ffmpeg}, if the format is not specified with the
+Note that with @file{ffmpeg}, if the format is not specified with the
@code{-f} option and the output filename specifies an image file
 format, the image2 muxer is automatically selected, so the previous
 command can be written as:
@example
-ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg'
+ffmpeg -i in.avi -r 1 'img-%03d.jpeg'
@end example

 Note also that the pattern must not necessarily contain "%d" or
 "%0@var{N}d", for example to create a single image file
@file{img.jpeg} from the input video you can employ the command:
@example
-ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
+ffmpeg -i in.avi -f image2 -vframes 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
-
-MD5 testing format.
-
-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:
-
-@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.
@@ -401,7 +177,7 @@ and @code{service_name}. If they are not set the default for
@code{service_name} is "Service01".

@example
-ffmpeg -i file.mpg -c copy \
+ffmpeg -i file.mpg -acodec copy -vcodec copy \
     -mpegts_original_network_id 0x1122 \
     -mpegts_transport_stream_id 0x3344 \
     -mpegts_service_id 0x5566 \
@@ -419,14 +195,14 @@ Null muxer.
 This muxer does not generate any output file, it is mainly useful for
 testing or benchmarking purposes.

-For example to benchmark decoding with @command{ffmpeg} you can use the
+For example to benchmark decoding with @file{ffmpeg} you can use the
 command:
@example
 ffmpeg -benchmark -i INPUT -f null out.null
@end example

 Note that the above command does not read or write the @file{out.null}
-file, but specifying the output file is required by the @command{ffmpeg}
+file, but specifying the output file is required by the @file{ffmpeg}
 syntax.

 Alternatively you can write the command as:
@@ -494,245 +270,7 @@ Both eyes laced in one Block, Right-eye view is first

 For example a 3D WebM clip can be created using the following command line:
@example
-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
-
-Basic stream segmenter.
-
-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.
-
-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:
-
-@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_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}.
-
-@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.
-@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
-@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
+ffmpeg -i sample_left_right_clip.mpg -an -vcodec libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
@end example

@c man end MUXERS
--- a/doc/nut.texi
+++ b/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
-
--- a/doc/optimization.txt
+++ b/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:
--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -22,95 +22,13 @@ 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.

@section sdl

-SDL (Simple DirectMedia Layer) output device.
+SDL (Simple Directmedia Layer) output device.

 This output devices allows to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
@@ -137,13 +55,12 @@ 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

-The following command shows the @command{ffmpeg} output is an
+The following command shows the @file{ffmpeg} output is an
 SDL window, forcing its size to the qcif format:
@example
 ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -1,370 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Platform Specific Information
-@titlepage
-@center @titlefont{Platform Specific Information}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Unix-like
-
-Some parts of FFmpeg cannot be built with version 2.15 of the GNU
-assembler which is still provided by a few AMD64 distributions. To
-make sure your compiler really uses the required version of gas
-after a binutils upgrade, run:
-
-@example
-$(gcc -print-prog-name=as) --version
-@end example
-
-If not, then you should install a different compiler that has no
-hard-coded path to gas. In the worst case pass @code{--disable-asm}
-to configure.
-
-@section BSD
-
-BSD make will not build FFmpeg, you need to install and use GNU Make
-(@command{gmake}).
-
-@section (Open)Solaris
-
-GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}),
-standard Solaris Make will not work. When building with a non-c99 front-end
-(gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
-or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
-since the libc is not c99-compliant by default. The probes performed by
-configure may raise an exception leading to the death of configure itself
-due to a bug in the system shell. Simply invoke a different shell such as
-bash directly to work around this:
-
-@example
-bash ./configure
-@end example
-
-@anchor{Darwin}
-@section Darwin (Mac OS X, iPhone)
-
-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
-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
-optimized assembler functions. @uref{http://www.finkproject.org/, Fink},
-@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix},
-@uref{http://mxcl.github.com/homebrew/, Homebrew}
-or @uref{http://www.macports.org, MacPorts} can easily provide it.
-
-
-@chapter DOS
-
-Using a cross-compiler is preferred for various reasons.
-@url{http://www.delorie.com/howto/djgpp/linux-x-djgpp.html}
-
-
-@chapter OS/2
-
-For information about compiling FFmpeg on OS/2 see
-@url{http://www.edm2.com/index.php/FFmpeg}.
-
-
-@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/}.
-
-@section Native Windows compilation using MinGW or MinGW-w64
-
-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.
-
-Notes:
-
-@itemize
-
-@item Building natively using MSYS can be sped up by disabling implicit rules
-in the Makefile by calling @code{make -r} instead of plain @code{make}. This
-speed up is close to non-existent for normal one-off builds and is only
-noticeable when running make for a second time (for example during
-@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.
-
-@item By using @code{./configure --enable-shared} when configuring FFmpeg,
-you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
-libavformat) as DLLs.
-
-@end itemize
-
-@section Microsoft Visual C++
-
-FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and
-wrapper.
-
-You will need the following prerequisites:
-
-@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
-
-To set up a proper MSVC environment in MSYS, you simply need to run
-@code{msys.bat} from the Visual Studio command prompt.
-
-Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
-somewhere in your @code{PATH}.
-
-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.
-
-Finally, run:
-
-@example
-./configure --toolchain=msvc
-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.
-
-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.
-
-@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.
-
-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
-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}.
-This issue is reported upstream at
-@url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}.
-
-To create import libraries that work with the @code{/OPT:REF} option
-(which is enabled by default in Release mode), follow these steps:
-
-@enumerate
-
-@item Open the @emph{Visual Studio 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}).
-
-@item Enter the @file{bin} directory where the created LIB and DLL files
-are stored.
-
-@item Generate new import libraries with @command{lib.exe}:
-
-@example
-lib /machine:i386 /def:..\lib\foo-version.def  /out:foo.lib
-@end example
-
-Replace @code{foo-version} and @code{foo} with the respective library names.
-
-@end enumerate
-
-@anchor{Cross compilation for Windows with Linux}
-@section Cross compilation for Windows with Linux
-
-You must use the MinGW cross compilation tools available at
-@url{http://www.mingw.org/}.
-
-Then configure FFmpeg with the following options:
-@example
-./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
-@end example
-(you can change the cross-prefix according to the prefix chosen for the
-MinGW tools).
-
-Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}.
-
-@section Compilation under Cygwin
-
-Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
-llrint() in its C library.
-
-Install your Cygwin with all the "Base" packages, plus the
-following "Devel" ones:
-@example
-binutils, gcc4-core, make, git, mingw-runtime, texi2html
-@end example
-
-In order to run FATE you will also need the following "Utils" packages:
-@example
-bc, diffutils
-@end example
-
-If you want to build FFmpeg with additional libraries, download Cygwin
-"Devel" packages for Ogg and Vorbis from any Cygwin packages repository:
-@example
-libogg-devel, libvorbis-devel
-@end example
-
-These library packages are only available from
-@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
-@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.
-
-@section Crosscompilation for Windows under Cygwin
-
-With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
-
-Just install your Cygwin as explained before, plus these additional
-"Devel" packages:
-@example
-gcc-mingw-core, mingw-runtime, mingw-zlib
-@end example
-
-and add some special flags to your configure invocation.
-
-For a static build run
-@example
-./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
-@end example
-
-and for a build with shared libraries
-@example
-./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
--- a/doc/print_options.c
+++ b/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;
-}
--- a/doc/protocols.texi
+++ b/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
@@ -66,7 +52,7 @@ resource to be concatenated, each one possibly specifying a distinct
 protocol.

 For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
+@file{split2.mpeg}, @file{split3.mpeg} with @file{ffplay} use the
 command:
@example
 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@@ -75,22 +61,13 @@ 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.

 Allow to read from or read to a file.

-For example to read from a file @file{input.mpeg} with @command{ffmpeg}
+For example to read from a file @file{input.mpeg} with @file{ffmpeg}
 use the command:
@example
 ffmpeg -i file:input.mpeg output.mpeg
@@ -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).
@@ -177,14 +134,14 @@ pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{number}
 is not specified, by default the stdout file descriptor will be used
 for writing, stdin for reading.

-For example to read from stdin with @command{ffmpeg}:
+For example to read from stdin with @file{ffmpeg}:
@example
 cat test.wav | ffmpeg -i pipe:0
 # ...this is the same as...
 cat test.wav | ffmpeg -i pipe:
@end example

-For writing to stdout with @command{ffmpeg}:
+For writing to stdout with @file{ffmpeg}:
@example
 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
 # ...this is the same as...
@@ -198,12 +155,12 @@ be seekable, so they will fail with the pipe output protocol.

 Real-Time Messaging Protocol.

-The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
-content across a TCP/IP network.
+The Real-Time Messaging Protocol (RTMP) is used for streaming multime‐
+dia 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:
@@ -218,144 +175,27 @@ 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

-For example to read with @command{ffplay} a multimedia resource named
+For example to read with @file{ffplay} a multimedia resource named
 "sample" from the application "vod" from an RTMP server "myserver":
@example
 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
 librtmp.

 Requires the presence of the librtmp headers and library during
-configuration. You need to explicitly configure the build with
+configuration. You need to explicitely configure the build with
 "--enable-librtmp". If enabled this will replace the native RTMP
 protocol.

@@ -379,12 +219,12 @@ meaning as specified for the RTMP native protocol.
 See the librtmp manual page (man 3 librtmp) for more information.

 For example, to stream a file in real-time to an RTMP server using
-@command{ffmpeg}:
+@file{ffmpeg}:
@example
 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
@end example

-To play the same stream using @command{ffplay}:
+To play the same stream using @file{ffplay}:
@example
 ffplay "rtmp://myserver/live/mystream live=1"
@end example
@@ -402,19 +242,16 @@ data transferred over RDT).

 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
-@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
+RTSP server, @url{http://github.com/revmischa/rtsp-server}).

 The required syntax for a RTSP url is:
@example
-rtsp://@var{hostname}[:@var{port}]/@var{path}
+rtsp://@var{hostname}[:@var{port}]/@var{path}[?@var{options}]
@end example

-The following options (set on the @command{ffmpeg}/@command{ffplay} command
-line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
+@var{options} is a @code{&}-separated list. The following options
 are supported:

-Flags for @code{rtsp_transport}:
-
@table @option

@item udp
@@ -424,33 +261,27 @@ Use UDP as lower transport protocol.
 Use TCP (interleaving within the RTSP control channel) as lower
 transport protocol.

-@item udp_multicast
+@item multicast
 Use UDP multicast as lower transport protocol.

@item http
 Use HTTP tunneling as lower transport protocol, which is useful for
 passing proxies.
+
+@item filter_src
+Accept packets only from negotiated peer address and port.
@end table

 Multiple lower transport protocols may be specified, in that case they are
 tried one at a time (if the setup of one fails, the next one is tried).
 For the muxer, only the @code{tcp} and @code{udp} options are supported.

-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
+When watching multi-bitrate Real-RTSP streams with @file{ffplay}, the
 streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.
@@ -460,13 +291,13 @@ Example command lines:
 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:

@example
-ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
+ffplay -max_delay 500000 rtsp://server/video.mp4?udp
@end example

 To watch a stream tunneled over HTTP:

@example
-ffplay -rtsp_transport http rtsp://server/video.mp4
+ffplay rtsp://server/video.mp4?http
@end example

 To send a stream in realtime to a RTSP server, for others to watch:
@@ -475,12 +306,6 @@ 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
@@ -533,13 +358,13 @@ To broadcast a stream on the local subnet, for watching in VLC:
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example

-Similarly, for watching in @command{ffplay}:
+Similarly, for watching in ffplay:

@example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
@end example

-And for watching in @command{ffplay}, over IPv6:
+And for watching in ffplay, over IPv6:

@example
 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
@@ -587,11 +412,6 @@ 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 +419,6 @@ 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.
@@ -650,37 +428,25 @@ 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.
-
-@item localaddr=@var{addr}
-Choose the local IP address. This is useful e.g. if sending multicast
-and the host has multiple interfaces, where the user can choose
-which interface to send on by specifying the IP address of that interface.
+override the local UDP port to bind with

@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 +458,9 @@ 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 @file{ffmpeg} follow.

 To stream over UDP to a remote endpoint:
@example
--- a/doc/rate_distortion.txt
+++ b/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
--- a/doc/soc.txt
+++ b/doc/soc.txt
@@ -18,7 +18,7 @@ essential that changes to their codebase are publicly visible, clean and
 easy reviewable that again leads us to:
 * use of a revision control system like git
 * separation of cosmetic from non-cosmetic changes (this is almost entirely
-  ignored by mentors and students in soc 2006 which might lead to a surprise
+  ignored by mentors and students in soc 2006 which might lead to a suprise
  when the code will be reviewed at the end before a possible inclusion in
  FFmpeg, individual changes were generally not reviewable due to cosmetics).
 * frequent commits, so that comments can be provided early
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .1.14
 .7