h263: disable loop filter with lowres

Fixes ticket1212

Found-by: Piotr Bandurski <ami_stuff@o2.pl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cc229d4e83)

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

.gitignore

@@ -1,68 +1,45 @@
-*.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/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

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!
+
+

CREDITS

@@ -1,7 +1,5 @@
-This file contains the names of some of the people who have contributed to
-FFmpeg. The names are sorted alphabetically by last name.  As this file is
-currently quite outdated and git serves as a much better tool for determining
-authorship, it remains here for historical reasons only.
+This file contains the name of the people who have contributed to
+FFmpeg. The names are sorted alphabetically by last name.
 
 Dénes Balatoni
 Michel Bardiaux

Changelog

@@ -0,0 +1,741 @@
+Entries are sorted chronologically from oldest to youngest within each release,
+releases are sorted from youngest to oldest.
+
+version 0.6.6:
+- id3v2: fix skipping extended header in id3v2.4
+- nsvdec: Fix use of uninitialized streams
+- nsvdec: Be more careful with av_malloc()
+- nsvdec: Propagate errors
+- dv: check stype
+- dv: Fix null pointer dereference due to ach=0
+- dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
+- atrac3: Fix crash in tonal component decoding
+- mjpegbdec: Fix overflow in SOS
+- kgv1dec: Increase offsets array size so it is large enough
+- motionpixels: decode only the 111 complete frames for fate
+- motionpixels: Clip YUV values after applying a gradient
+- celp filters: Do not read earlier than the start of the 'out' vector
+- vqavideo: return error if image size is not a multiple of block size (CVE-2012-0947)
+- dpcm: ignore extra unpaired bytes in stereo streams (CVE-2011-3951)
+- aacsbr: prevent out of bounds memcpy() (CVE-2012-0850)
+- h264: Add check for invalid chroma_format_idc (CVE-2012-0851)
+- adpcm: ADPCM Electronic Arts has always two channels (CVE-2012-0852)
+- shorten: remove VLA and check for buffer overflow
+- shorten: Prevent block size from increasing
+- shorten: Fix out of bound writes in fix_bitshift()
+- shorten: check for realloc failure (CVE-2012-0858)
+- shorten: Use separate pointers for the allocated memory for decoded samples (CVE-2012-0858)
+- kmvc: Check palsize (CVE-2011-3952)
+- qdm2: clip array indices returned by qdm2_get_vlc()
+- configure: properly check for mingw-w64 through installed headers
+- mingw32: properly check if vfw capture is supported by the system headers
+- mingw32: merge checks for mingw-w64 and mingw32-runtime >= 3.15 into one
+- dxva2: define required feature selection macros
+- png: check bit depth for PAL8/Y400A pixel formats
+- ea: check chunk_size for validity
+- tqi: Pass errors from the MB decoder
+
+
+version 0.6.5:
+- vorbis: An additional defense in the Vorbis codec. (CVE-2011-3895)
+- vorbisdec: Fix decoding bug with channel handling.
+- matroskadec: Fix a bug where a pointer was cached to an array that might
+  later move due to a realloc(). (CVE-2011-3893)
+- vorbis: Avoid some out-of-bounds reads. (CVE-2011-3893)
+- vp3: fix oob read for negative tokens and memleaks on error, (CVE-2011-3892)
+- vp3: fix streams with non-zero last coefficient.
+
+
+version 0.6.4:
+- 4xm: Add a check in decode_i_frame to prevent buffer overreads
+- wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
+- swscale: #include "libavutil/mathematics.h"
+- vp3dec: Check coefficient index in vp3_dequant() (CVE-2011-4352)
+- svq1dec: call avcodec_set_dimensions() after dimensions changed. (CVE-2011-4579)
+- commits related to CVE-2011-4353:
+  - vp6: Fix illegal read.
+  - vp6: Reset the internal state when aborting key frames header parsing
+  - vp6: Check for huffman tree build errors
+  - vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
+- commits related to CVE-2011-4351:
+  - qdm2: check output buffer size before decoding
+  - Fix qdm2 decoder packet handling to match the api
+  - Fix out of bound reads in the QDM2 decoder.
+  - Check for out of bound writes in the QDM2 decoder.
+- vmd: fix segfaults on corruped streams (CVE-2011-4364)
+- rv34: Check for invalid slice offsets
+- rv34: Fix potential overreads
+- rv34: Avoid NULL dereference on corrupted bitstream
+- rv10: Reject slices that does not have the same type as the first one
+- oggdec: fix out of bound write in the ogg demuxer
+- smacker: fix a few off by 1 errors
+- Check for invalid VLC value in smacker decoder.
+- Check and propagate errors when VLC trees cannot be built in smacker decoder.
+- Fixed off by one packet size allocation in the smacker demuxer.
+- Check for invalid packet size in the smacker demuxer.
+- ape demuxer: fix segfault on memory allocation failure.
+- Fixed size given to init_get_bits() in xan decoder.
+- smacker demuxer: handle possible av_realloc() failure.
+- Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
+- indeo2: fail if input buffer too small
+- indeo2: init_get_bits size in bits instead of bytes
+- wavpack: Check error codes rather than working around error conditions.
+- Fixed invalid writes and reads in wavpack decoder on corrupted bitstreams.
+- cpu detection: avoid a signed overflow
+- h264: correct implicit weight table computation for long ref pics
+- h264: correct the check for invalid long term frame index in MMCO decode
+- rv10/20: tell decoder to use edge emulation
+- flvenc: use int64_t to store offsets
+- VC-1: fix reading of custom PAR.
+- h264: notice memory allocation failure
+- libx264: do not set pic quality if no frame is output
+- mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
+- rv30: return AVERROR(EINVAL) instead of EINVAL
+- Do not decode RV30 files if the extradata is too small
+- aacps: skip some memcpy() if src and dst would be equal
+- mpegts: fix Continuity Counter error detection
+- alsa: fallback to buffer_size/4 for period_size.
+- mxfenc: fix ignored drop flag in binary timecode representation.
+- dca: set AVCodecContext frame_size for DTS audio
+- H.264: fix overreads of qscale_table
+- postprocess.c: filter name needs to be double 0 terminated
+- Replace strncpy() with av_strlcpy() in libpostproc.
+- jpegdec: actually search for and parse RSTn
+- riff: Add mpgv MPEG-2 fourcc
+- Added M701 codec_tag for mpeg2video
+
+
+version 0.6.3:
+
+- fix compilation with --enable-hardcoded-tables
+- mjpeg: Detect overreads in mjpeg_decode_scan() and error out.
+- aac: add headers needed for log2f()
+- lavf: inspect more frames for fps when container time base is coarse
+- AMV: disable DR1 and don't override EMU_EDGE
+  (addresses http://seclists.org/bugtraq/2011/Apr/257)
+- Fix memory (re)allocation in matroskadec.c (MSVR11-011/CVE-2011-3504)
+- Fix some crashes with invalid bitstreams in the CAVS decoder
+  (CVE-2011-3362, CVE-2011-3973, CVE-2011-3974)
+- Compilation fixes for gcc-4.6, testsuite now passes again
+- Fix a heap corruption issue in the OGG decoder
+- Backported the Android VisualOn AAC encoder wrapper from 0.7.2
+
+
+version 0.6.3:
+
+- AMV: Fix possibly exploitable crash.
+- Fix apparently exploitable race condition.
+  (addresses http://seclists.org/bugtraq/2011/Apr/257)
+
+version 0.6.2:
+
+- fix compilation with --enable-hardcoded-tables
+- Fix invalid reads in VC-1 decoding (related to CVE-2011-0723)
+- Do not attempt to decode APE file with no frames
+  (adresses http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
+
+
+version 0.6.1:
+
+- fix autodetection of E-AC-3 substream samples
+- performance fix for seekable HTTP
+- backport AAC-HE v2 from trunk
+- add missing VP80 fourcc code for the VP8 codec
+- small documentation fixes
+- fix several potentially exploitable issues in the FLIC decoder
+  (addresses CVE-2010-3429)
+
+
+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
+- WebM support in Matroska de/muxer
+- low overhead Ogg muxing
+- VP8 de/encoding via libvpx
+- CODEC_CAP_EXPERIMENTAL added
+
+
+
+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
+
+
+
+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

INSTALL

@@ -9,7 +9,3 @@ path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
 2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
 
 3) Type 'make install' to install all binaries and libraries you built.
-
-NOTICE
-
- - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

LICENSE

@@ -1,4 +1,5 @@
 FFmpeg:
+-------
 
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
 or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
@@ -13,44 +14,18 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are
 
 - libpostproc
-- libmpcodecs
+- optional MMX optimizations for YUV to RGB colorspace conversion in
+  libswscale/x86/yuv2rgb_template.c
 - optional x86 optimizations in the files
+  libavcodec/x86/h264_deblock_sse2.asm
+  libavcodec/x86/h264_idct_sse2.asm
   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 +33,18 @@ 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.

MAINTAINERS

@@ -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
 ==============
@@ -25,10 +19,7 @@ ffmpeg:
   ffmpeg.c                              Michael Niedermayer
 
 ffplay:
-  ffplay.c                              Marton Balint
-
-ffprobe:
-  ffprobe.c                             Stefano Sabatini
+  ffplay.c                              Michael Niedermayer
 
 ffserver:
   ffserver.c, ffserver.h                Baptiste Coudurier
@@ -43,14 +34,14 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Mike Melanson
-website                                 Robert Swain, Lou Logan
+documentation                           Mike Melanson, Diego Biurrun
+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 +78,6 @@ 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:
@@ -109,19 +98,17 @@ Generic Parts:
     simple_idct.c, simple_idct.h        Michael Niedermayer
   postprocessing:
     libpostproc/*                       Michael Niedermayer
-  table generation:
-    tableprint.c, tableprint.h          Reimar Doeffinger
 
 Codecs:
   4xm.c                                 Michael Niedermayer
   8bps.c                                Roberto Togni
   8svx.c                                Jaikrishnan Menon
   aasc.c                                Kostya Shishkov
+  aac*, sbr.h                           Alex Converse
   ac3*                                  Justin Ruggles
   alacenc.c                             Jaikrishnan Menon
   alsdec.c                              Thilo Borgmann
   apedec.c                              Kostya Shishkov
-  ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3*                               Benjamin Larsson
   bgmc.c, bgmc.h                        Thilo Borgmann
@@ -130,13 +117,9 @@ 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
   dnxhd*                                Baptiste Coudurier
@@ -145,11 +128,9 @@ 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
-  g722.c                                Martin Storsjo
   g726.c                                Roman Shaposhnik
   gifdec.c                              Baptiste Coudurier
   h264*                                 Loren Merritt, Michael Niedermayer
@@ -162,28 +143,21 @@ 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
   loco.c                                Kostya Shishkov
   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
@@ -194,9 +168,7 @@ 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
   qcelp*                                Reynaldo H. Verdejo Pinochet
   qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
@@ -216,10 +188,8 @@ Codecs:
   smc.c                                 Mike Melanson
   snow.c                                Michael Niedermayer, Loren Merritt
   sonic.c                               Alex Beregszaszi
-  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
@@ -229,20 +199,15 @@ 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
+  vorbis_dec.c                          Denes Balatoni
   vp3*                                  Mike Melanson
   vp5                                   Aurelien Jacobs
   vp6                                   Aurelien Jacobs
-  vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
   vqavideo.c                            Mike Melanson
   wavpack.c                             Kostya Shishkov
   wmaprodec.c                           Sascha Sommer
@@ -250,19 +215,13 @@ 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 +231,10 @@ 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 +249,15 @@ 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
@@ -333,25 +265,17 @@ Muxers/Demuxers:
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
   idcin.c                               Mike Melanson
-  idroqdec.c                            Mike Melanson
+  idroq.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
@@ -362,69 +286,42 @@ 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
   oggdec.c, oggdec.h                    David Conrad
   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
   rl2.c                                 Sascha Sommer
   rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
   rtmp*                                 Kostya Shishkov
-  rtp.c, rtpenc.c                       Martin Storsjo
-  rtpdec_asf.*                          Ronald S. Bultje
-  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
+  rtp.c, rtpenc.c                       Luca Abeni
+  rtp_asf.*                             Ronald S. Bultje
+  rtp_mpv.*, rtp_aac.*                  Luca Abeni
   rtsp.c                                Luca Barbato
-  sbgdec.c                              Nicolas George
-  sdp.c                                 Martin Storsjo
+  sdp.c                                 Luca Abeni
   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
   wav.c                                 Michael Niedermayer
   wc3movie.c                            Mike Melanson
   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
 =====================================
 
@@ -432,6 +329,7 @@ Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
 MIPS                                    Mans Rullgard
+BeOS                                    Francois Revol
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
@@ -442,43 +340,20 @@ Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer
 
 
-Releases
-========
+GnuPG Fingerprints of maintainers and others who have svn write access
+======================================================================
 
-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
-==================================================
-
-Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
-Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
 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

Makefile

@@ -1,80 +1,62 @@
-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 %.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
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher
-TOOLS-$(CONFIG_ZLIB) += cws2fws
+PROGS      := $(addsuffix   $(EXESUF), $(PROGS-yes))
+PROGS_G     = $(addsuffix _g$(EXESUF), $(PROGS-yes))
+OBJS        = $(addsuffix .o,          $(PROGS-yes)) cmdutils.o
+MANPAGES    = $(addprefix doc/, $(addsuffix .1, $(PROGS-yes)))
+TOOLS       = $(addprefix tools/, $(addsuffix $(EXESUF), cws2fws pktdumper probetest qt-faststart trasher))
+HOSTPROGS   = $(addprefix tests/, audiogen videogen rotozoom tiny_psnr)
 
 BASENAMES   = ffmpeg ffplay ffprobe ffserver
-ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-ALLMANPAGES = $(BASENAMES:%=%.1)
+ALLPROGS    = $(addsuffix   $(EXESUF), $(BASENAMES))
+ALLPROGS_G  = $(addsuffix _g$(EXESUF), $(BASENAMES))
+ALLMANPAGES = $(addsuffix .1, $(BASENAMES))
 
 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: $(PROGS)
+ALL_TARGETS-$(CONFIG_DOC)       += documentation
+
+ifdef PROGS
+INSTALL_TARGETS-yes             += install-progs install-data
+INSTALL_TARGETS-$(CONFIG_DOC)   += install-man
+endif
+INSTALL_PROGS_TARGETS-$(CONFIG_SHARED) = install-libs
+
+all: $(FF_DEP_LIBS) $(PROGS) $(ALL_TARGETS-yes)
 
 $(PROGS): %$(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))
-	@-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
 
 define RESET
 $(1) :=
@@ -84,35 +66,30 @@ 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-$(1):.o=.d)
-endef
+ffplay_g$(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) $(OBJS-$*) $(FF_EXTRALIBS)
+tools/%$(EXESUF): tools/%.o
+	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
 
-OBJDIRS += tools
+tools/%.o: tools/%.c
+	$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_O) $<
 
--include $(wildcard tools/*.d)
+ffplay.o ffplay.d: CFLAGS += $(SDL_CFLAGS)
 
-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
+SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
 
-.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
+.version: $(wildcard $(GIT_LOG) $(SVN_ENTRIES)) $(VERSION_SH) config.mak
 .version: M=@
 
 version.h .version:
@@ -122,27 +99,39 @@ version.h .version:
 # force version.sh to run whenever version might have changed
 -include .version
 
-ifdef PROGS
-install: install-progs install-data
-endif
+alltools: $(TOOLS)
 
-install: install-libs install-headers
+documentation: $(addprefix doc/, developer.html faq.html ffmpeg-doc.html \
+                                 ffplay-doc.html ffprobe-doc.html ffserver-doc.html \
+                                 general.html libavfilter.html $(ALLMANPAGES))
 
-install-libs: install-libs-yes
+doc/%.html: TAG = HTML
+doc/%.html: doc/%.texi
+	$(M)cd doc && texi2html -monolithic -number $(<:doc/%=%)
 
-install-progs-yes:
-install-progs-$(CONFIG_SHARED): install-libs
+doc/%.pod: TAG = POD
+doc/%.pod: doc/%-doc.texi
+	$(M)doc/texi2pod.pl $< $@
 
-install-progs: install-progs-yes $(PROGS)
+doc/%.1: TAG = MAN
+doc/%.1: doc/%.pod
+	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
+
+install: $(INSTALL_TARGETS-yes)
+
+install-progs: $(PROGS) $(INSTALL_PROGS_TARGETS-yes)
 	$(Q)mkdir -p "$(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-progs uninstall-data uninstall-man
 
 uninstall-progs:
 	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
@@ -150,44 +139,215 @@ 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) $(addprefix tests/,$(addsuffix $(HOSTEXESUF),audiogen videogen rotozoom tiny_psnr))
+
+clean:: testclean
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(CLEANSUFFIXES:%=tools/%)
-	$(RM) coverage.info
-	$(RM) -r coverage-html
+	$(RM) doc/*.html doc/*.pod doc/*.1
+	$(RM) $(TOOLS)
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .version version.h libavutil/avconfig.h libavcodec/codec_names.h
+	$(RM) version.h config.* 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 checkheaders
 
-check: all alltools examples testprogs fate
+fulltest test: codectest lavftest seektest
 
-include $(SRC_PATH)/doc/Makefile
-include $(SRC_PATH)/tests/Makefile
+FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
+SEEK_REFFILE     = $(SRC_PATH)/tests/seek.regression.ref
 
-$(sort $(OBJDIRS)):
-	$(Q)mkdir -p $@
+ENCDEC = $(and $(CONFIG_$(1)_ENCODER),$(CONFIG_$(1)_DECODER))
+MUXDEM = $(and $(CONFIG_$(1)_MUXER),$(CONFIG_$(or $(2),$(1))_DEMUXER))
 
-# Dummy rule to stop make trying to rebuild removed or renamed headers
-%.h:
-	@:
+VCODEC_TESTS =
+VCODEC_TESTS-$(call ENCDEC,ASV1)             += asv1
+VCODEC_TESTS-$(call ENCDEC,ASV2)             += asv2
+VCODEC_TESTS-$(call ENCDEC,DNXHD)            += dnxhd_1080i dnxhd_720p dnxhd_720p_rd
+VCODEC_TESTS-$(call ENCDEC,DVVIDEO)          += dv dv50
+VCODEC_TESTS-$(call ENCDEC,FFV1)             += ffv1
+VCODEC_TESTS-$(call ENCDEC,FLASHSV)          += flashsv
+VCODEC_TESTS-$(call ENCDEC,FLV)              += flv
+VCODEC_TESTS-$(call ENCDEC,H261)             += h261
+VCODEC_TESTS-$(call ENCDEC,H263)             += h263 h263p
+VCODEC_TESTS-$(call ENCDEC,HUFFYUV)          += huffyuv
+VCODEC_TESTS-$(call ENCDEC,JPEGLS)           += jpegls
+VCODEC_TESTS-$(call ENCDEC,MJPEG)            += mjpeg ljpeg
+VCODEC_TESTS-$(call ENCDEC,MPEG1VIDEO)       += mpeg mpeg1b
+VCODEC_TESTS-$(call ENCDEC,MPEG2VIDEO)       += mpeg2 mpeg2thread
+VCODEC_TESTS-$(call ENCDEC,MPEG4)            += mpeg4 mpeg4adv mpeg4nr mpeg4thread error rc
+VCODEC_TESTS-$(call ENCDEC,MSMPEG4V1)        += msmpeg4
+VCODEC_TESTS-$(call ENCDEC,MSMPEG4V2)        += msmpeg4v2
+VCODEC_TESTS-$(call ENCDEC,ROQ)              += roq
+VCODEC_TESTS-$(call ENCDEC,RV10)             += rv10
+VCODEC_TESTS-$(call ENCDEC,RV20)             += rv20
+VCODEC_TESTS-$(call ENCDEC,SNOW)             += snow snowll
+VCODEC_TESTS-$(call ENCDEC,SVQ1)             += svq1
+VCODEC_TESTS-$(call ENCDEC,WMV1)             += wmv1
+VCODEC_TESTS-$(call ENCDEC,WMV2)             += wmv2
 
-# Disable suffix rules.  Most of the builtin rules are suffix rules,
-# so this saves some time on slow systems.
-.SUFFIXES:
+ACODEC_TESTS =
+ACODEC_TESTS-$(call ENCDEC,AC3)              += ac3
+ACODEC_TESTS-$(call ENCDEC,ADPCM_G726)       += g726
+ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_QT)     += adpcm_ima_qt
+ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_WAV)    += adpcm_ima_wav
+ACODEC_TESTS-$(call ENCDEC,ADPCM_MS)         += adpcm_ms
+ACODEC_TESTS-$(call ENCDEC,ADPCM_SWF)        += adpcm_swf
+ACODEC_TESTS-$(call ENCDEC,ADPCM_YAMAHA)     += adpcm_yam
+ACODEC_TESTS-$(call ENCDEC,ALAC)             += alac
+ACODEC_TESTS-$(call ENCDEC,FLAC)             += flac
+ACODEC_TESTS-$(call ENCDEC,MP2)              += mp2
+ACODEC_TESTS-$(call ENCDEC,PCM_S16LE)        += pcm         # fixme
+ACODEC_TESTS-$(call ENCDEC,WMAV1)            += wmav1
+ACODEC_TESTS-$(call ENCDEC,WMAV1)            += wmav2
 
-.PHONY: all all-yes alltools check *clean config install*
-.PHONY: testprogs uninstall*
+LAVF_TESTS =
+LAVF_TESTS-$(call MUXDEM,AIFF)               += aiff
+LAVF_TESTS-$(call MUXDEM,PCM_ALAW)           += alaw
+LAVF_TESTS-$(call MUXDEM,ASF)                += asf
+LAVF_TESTS-$(call MUXDEM,AU)                 += au
+LAVF_TESTS-$(call MUXDEM,AVI)                += avi
+LAVF_TESTS-$(call ENCDEC,BMP)                += bmp
+LAVF_TESTS-$(call MUXDEM,DV)                 += dv_fmt
+LAVF_TESTS-$(call MUXDEM,FFM)                += ffm
+LAVF_TESTS-$(call MUXDEM,FLV)                += flv_fmt
+LAVF_TESTS-$(call ENCDEC,GIF)                += gif
+LAVF_TESTS-$(call MUXDEM,GXF)                += gxf
+LAVF_TESTS-$(call ENCDEC,MJPEG)              += jpg
+LAVF_TESTS-$(call MUXDEM,MATROSKA)           += mkv
+LAVF_TESTS-$(call MUXDEM,MMF)                += mmf
+LAVF_TESTS-$(call MUXDEM,MOV)                += mov
+LAVF_TESTS-$(call MUXDEM,MPEG1SYSTEM,MPEGPS) += mpg
+LAVF_TESTS-$(call MUXDEM,PCM_MULAW)          += mulaw
+LAVF_TESTS-$(call MUXDEM,MXF)                += mxf
+LAVF_TESTS-$(call MUXDEM,NUT)                += nut
+LAVF_TESTS-$(call MUXDEM,OGG)                += ogg
+LAVF_TESTS-$(call ENCDEC,PBM)                += pbmpipe
+LAVF_TESTS-$(call ENCDEC,PCX)                += pcx
+LAVF_TESTS-$(call ENCDEC,PGM)                += pgm pgmpipe
+LAVF_TESTS-$(call MUXDEM,RAWVIDEO)           += pixfmt
+LAVF_TESTS-$(call ENCDEC,PPM)                += ppm ppmpipe
+LAVF_TESTS-$(call MUXDEM,RM)                 += rm
+LAVF_TESTS-$(call ENCDEC,SGI)                += sgi
+LAVF_TESTS-$(call MUXDEM,SWF)                += swf
+LAVF_TESTS-$(call ENCDEC,TARGA)              += tga
+LAVF_TESTS-$(call ENCDEC,TIFF)               += tiff
+LAVF_TESTS-$(call MUXDEM,MPEGTS)             += ts
+LAVF_TESTS-$(call MUXDEM,VOC)                += voc
+LAVF_TESTS-$(call MUXDEM,WAV)                += wav
+LAVF_TESTS-$(call MUXDEM,YUV4MPEGPIPE)       += yuv4mpeg
+
+LAVFI_TESTS =           \
+    crop                \
+    crop_scale          \
+    crop_scale_vflip    \
+    crop_vflip          \
+    null                \
+    scale200            \
+    scale500            \
+    vflip               \
+    vflip_crop          \
+    vflip_vflip         \
+
+ACODEC_TESTS := $(addprefix regtest-, $(ACODEC_TESTS) $(ACODEC_TESTS-yes))
+VCODEC_TESTS := $(addprefix regtest-, $(VCODEC_TESTS) $(VCODEC_TESTS-yes))
+LAVF_TESTS  := $(addprefix regtest-, $(LAVF_TESTS)  $(LAVF_TESTS-yes))
+LAVFI_TESTS := $(addprefix regtest-, $(LAVFI_TESTS) $(LAVFI_TESTS-yes))
+
+CODEC_TESTS = $(VCODEC_TESTS) $(ACODEC_TESTS)
+
+codectest: $(CODEC_TESTS)
+lavftest:  $(LAVF_TESTS)
+lavfitest: $(LAVFI_TESTS)
+
+$(ACODEC_TESTS): regtest-aref
+$(VCODEC_TESTS): regtest-vref
+$(LAVF_TESTS) $(LAVFI_TESTS): regtest-ref
+
+REFFILE = $(SRC_PATH)/tests/ref/$(1)/$(2:regtest-%=%)
+RESFILE = tests/data/$(2:regtest-%=%).$(1).regression
+
+define CODECTEST_CMD
+	$(SRC_PATH)/tests/codec-regression.sh $@ vsynth1 tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	$(SRC_PATH)/tests/codec-regression.sh $@ vsynth2 tests/vsynth2 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+endef
+
+regtest-ref: regtest-aref regtest-vref
+
+regtest-vref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm
+	$(CODECTEST_CMD)
+
+regtest-aref: ffmpeg$(EXESUF) tests/data/asynth1.sw
+	@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
+
+$(VCODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
+	@echo "TEST VCODEC $(@:regtest-%=%)"
+	@$(CODECTEST_CMD)
+	@diff -u -w $(call REFFILE,vsynth1,$@) $(call RESFILE,vsynth1,$@)
+	@diff -u -w $(call REFFILE,vsynth2,$@) $(call RESFILE,vsynth2,$@)
+
+$(ACODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
+	@echo "TEST ACODEC $(@:regtest-%=%)"
+	@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	@diff -u -w $(call REFFILE,acodec,$@) $(call RESFILE,acodec,$@)
+
+$(LAVF_TESTS):
+	@echo "TEST LAVF  $(@:regtest-%=%)"
+	@$(SRC_PATH)/tests/lavf-regression.sh $@ lavf tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	@diff -u -w $(call REFFILE,lavf,$@) $(call RESFILE,lavf,$@)
+
+$(LAVFI_TESTS):
+	@echo "TEST LAVFI $(@:regtest-%=%)"
+	@$(SRC_PATH)/tests/lavfi-regression.sh $@ lavfi tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	@diff -u -w $(call REFFILE,lavfi,$@) $(call RESFILE,lavfi,$@)
+
+seektest: codectest lavftest tests/seek_test$(EXESUF)
+	$(SRC_PATH)/tests/seek-regression.sh $(SRC_PATH) "$(TARGET_EXEC)" "$(TARGET_PATH)"
+
+ffservertest: ffserver$(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
+	$(BUILD_ROOT)/$< 'tests/vsynth1/'
+
+tests/vsynth2/00.pgm: tests/rotozoom$(HOSTEXESUF)
+	mkdir -p tests/vsynth2
+	$(BUILD_ROOT)/$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
+
+tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF)
+	mkdir -p tests/data
+	$(BUILD_ROOT)/$< $@
+
+tests/seek_test$(EXESUF): tests/seek_test.o $(FF_DEP_LIBS)
+	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
+
+ifdef SAMPLES
+include $(SRC_PATH_BARE)/tests/fate.mak
+fate: $(FATE_TESTS)
+$(FATE_TESTS): ffmpeg$(EXESUF)
+	@echo "TEST FATE   $(@:fate-%=%)"
+	@$(SRC_PATH)/tests/fate-run.sh $@ "$(SAMPLES)" "$(TARGET_EXEC)" "$(TARGET_PATH)" '$(CMD)'
+else
+fate:
+	@echo "SAMPLES not specified, cannot run FATE"
+endif
+
+.PHONY: documentation *test regtest-* zlib-error alltools check config

README

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

RELEASE

@@ -1 +1,212 @@
-1.1.13
+Release Notes
+=============
+
+* 0.6 "Works with HTML5" June, 2010
+
+General notes
+-------------
+
+This release focuses on improvements for the new multimedia elements in HTML5.
+The H.264 and Theora decoders are now significantly faster, the Vorbis decoder
+has seen important updates and this release supports Google's newly released
+libvpx library for the VP8 codec and WebM container.
+
+Other important changes are additions of decoders including, but not limited to,
+Intel Indeo 5, WMA Pro, WMA Voice and HE-AAC.
+
+See the Changelog file for a list of significant changes.
+
+Please note that our policy on bug reports has not changed. We still only accept
+bug reports against HEAD of the FFmpeg trunk repository. If you are experiencing
+any issues with any formally released version of FFmpeg, please try a current
+version of the development code to check if the issue still exists. If it does,
+make your report against the development code following the usual bug reporting
+guidelines.
+
+
+API and other notable Changes
+-----------------------------
+
+Please see the file doc/APIchanges for programmer-centric information.
+
+Notable changes:
+- deprecated vhook subsystem removed
+- deprecated old scaler removed
+- nonfree libamr support for AMR-NB/WB decoding/encoding removed
+- RTMP support in libavformat
+- -formats option split into -formats, -codecs, -bsfs, and -protocols
+- ffprobe tool
+- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
+- CODEC_CAP_EXPERIMENTAL added
+
+
+Added Codecs:
+-------------
+
+- VQF demuxer
+- PCX encoder
+- CorePNG decoding support
+- 8088flex TMV demuxer and decoder
+- enable symbol versioning by default for linkers that support it
+- V210 decoder and encoder
+- 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
+- experimental AAC encoder
+- 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
+- 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 packetization and depacketization of H.263 and AMR
+- Bink demuxer and audio/video decoders
+- IFF PBM/ILBM bitmap decoder
+- Indeo 5 decoder
+- WMA Voice decoder
+- AMR-NB decoder
+- RTSP muxer
+- HE-AAC v1 decoder
+- Kega Game Video (KGV1) decoder
+- Psygnosis YOP demuxer and video decoder
+- RTP hinting in the mov/3gp/mp4 muxer
+- VP8 decoding via libvpx
+
+
+Notable license related changes
+-------------------------------
+
+- remaining GPL parts in AC-3 decoder converted to LGPL
+- libswscale can now be compiled in LGPL mode
+
+
+
+* 0.6.1
+
+General notes
+-------------
+
+This point release includes some updates to make the 0.6 release series usable
+for users that need to retain the existing behavior as closely as possible.
+The changes follow below:
+
+Bugfixes
+--------
+
+- fix autodetection of E-AC-3 substream samples
+- performance fix for seekable HTTP
+- add missing VP80 fourcc code for the VP8 codec
+- small documentation fixes
+- fix several potentially exploitable issues in the FLIC decoder
+  (addresses CVE-2010-3429)
+
+
+HE-AAC v2 backport
+------------------
+
+This release includes a backport of the AAC decoder from trunk, which
+enables proper playback of HE-AAC v2 media.
+
+
+* 0.6.2
+
+General notes
+-------------
+
+This is a maintenance-only release that addresses a small number of security
+and portability issues. Distributors and system integrators are encouraged
+to update and share their patches against this branch.
+
+Security fixes
+--------------
+
+Programming errors in container and codec implementations may lead to
+denial of service or the execution of arbitrary code if the user is
+tricked into opening a malformed media file or stream.
+
+Affected and updated have been the implementations of the following
+codecs and container formats:
+
+ - VC1 decoder (Change related to CVE-2011-0723)
+ - APE decoder (cf. http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
+
+
+* 0.6.3
+
+General notes
+-------------
+
+This is a mostly maintenance-only release that addresses a small number
+of bugs such as security and compilation issues. Moreover, this release
+has been updated to work with gcc-4.6 and the VisualOn AAC encoder has
+been backported from the Libav 0.7.2 release. Distributors and system
+integrators are encouraged to update and share their patches against
+this branch.
+For a full list of changes please see the Changelog file.
+
+
+* 0.6.4
+
+General notes
+-------------
+
+This mostly maintenance-only release that addresses a number a number of
+bugs such as security and compilation issues that have been brought to
+our attention. Among other (rather minor) fixes, this release features
+fixes for the QDM2 decoder (CVE-2011-4351), vp3 decoder (CVE-2011-4352),
+DoS in the VP5/VP6 decoders (CVE-2011-4353), a buffer overflow in the
+Sierra VMD decoder CVE-2011-4364, and a safety fix in the svq1 decoder
+(CVE-2011-4579).
+
+Distributors and system integrators are encouraged
+to update and share their patches against this branch.  For a full list
+of changes please see the Changelog file.
+
+
+* 0.6.5
+
+General notes
+-------------
+
+This mostly maintenance-only release that addresses a number a number of
+bugs such as security and compilation issues that have been brought to
+our attention. Among other (rather minor) fixes, this release features
+fixes for the VP3 decoder (CVE-2011-3892), vorbis decoder, and matroska
+demuxer (CVE-2011-3893 and CVE-2011-3895).
+
+Distributors and system integrators are encouraged
+to update and share their patches against this branch.  For a full list
+of changes please see the Changelog file.
+
+
+* 0.6.6
+
+General notes
+-------------
+
+This mostly maintenance-only release that addresses a number a number of
+bugs such as security and compilation issues that have been brought to
+our attention. Among other fixes, this release features includes
+security updates for the, DV decoder (CVE-2011-3929 and CVE-2011-3936),
+VQA Video Decoder (CVE-2012-0947), DPCM codecs (CVE-2011-3951), AAC SBR
+(CVE-2012-0850), H.264 (CVE-2012-0851), ADPCM (CVE-2012-0852), Shorten
+(CVE-2012-0858), and the KMVC decoder (CVE-2011-3952).
+
+Distributors and system integrators are encouraged
+to update and share their patches against this branch.  For a full list
+of changes please see the Changelog file or the git commit history.

VERSION

@@ -1 +1 @@
-1.1.13
+0.6.6

arch.mak

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

cmdutils.h

@@ -22,17 +22,11 @@
 #ifndef FFMPEG_CMDUTILS_H
 #define FFMPEG_CMDUTILS_H
 
-#include <stdint.h>
-
+#include <inttypes.h>
 #include "libavcodec/avcodec.h"
-#include "libavfilter/avfilter.h"
 #include "libavformat/avformat.h"
 #include "libswscale/swscale.h"
 
-#ifdef __MINGW32__
-#undef main /* We don't want SDL to override our main() */
-#endif
-
 /**
  * program name, defined by the program for show_version().
  */
@@ -43,81 +37,51 @@ 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;
-
-/**
- * Initialize the cmdutils option system, in particular
- * allocate the *_opts contexts.
- */
-void init_opts(void);
-/**
- * Uninitialize the cmdutils option system, in particular
- * free the *_opts contexts and their contents.
- */
-void uninit_opts(void);
-
-/**
- * Trivial log callback.
- * Only suitable for opt_help and similar since it lacks prefix handling.
- */
-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.
+ * Sets 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.
- * Exit from the application if the string cannot be correctly
+ * Parses a string and returns its corresponding value as a double.
+ * Exits from the application if the string cannot be correctly
  * 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
- * value as a number of microseconds. Exit from the application if
+ * Parses a string specifying a time and returns its corresponding
+ * value as a number of microseconds. Exits from the application if
  * 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 +89,9 @@ 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
@@ -148,225 +100,42 @@ 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
+#define OPT_FUNC2  0x0400
+#define OPT_INT64  0x0800
+#define OPT_EXIT   0x1000
      union {
-        void *dst_ptr;
-        int (*func_arg)(void *, const char *, const char *);
-        size_t off;
+        void (*func_arg)(const char *); //FIXME passing error code as int return would be nicer then exit() in the func
+        int *int_arg;
+        char **str_arg;
+        float *float_arg;
+        int (*func2_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);
+void show_help_options(const OptionDef *options, const char *msg, int mask, int value);
 
 /**
- * Show help for all options with given flags in class and all its
- * 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);
-
-/**
- * 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
+ * Parses the command line arguments.
  * @param options Array with the definitions required to interpret every
- * option of the form: -option_name [argument]
+ * 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,
+                   void (* parse_arg_function)(const char*));
+
+void set_context_opts(void *ctx, void *opts_ctx, int flags);
 
 /**
- * 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);
-
-/**
- * Print an error message to stderr, indicating filename and a human
+ * Prints an error message to stderr, indicating filename and a human
  * readable description of the error code err.
  *
  * If strerror_r() is not available the use of this function in a
@@ -376,208 +145,79 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
  */
 void print_error(const char *filename, int err);
 
+void list_fmts(void (*get_fmt_string)(char *buf, int buf_size, int fmt), int nb_fmts);
+
 /**
- * Print the program banner to stderr. The banner contents depend on the
+ * Prints the program banner to stderr. The banner contents depend on the
  * 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
+ * Prints 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
+ * Prints 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);
-
-/**
- * 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
+ * Prints a listing containing all the formats supported by the
  * program.
  */
-int show_decoders(void *optctx, const char *opt, const char *arg);
+void show_formats(void);
 
 /**
- * Print a listing containing all the encoders supported by the
+ * Prints a listing containing all the codecs 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);
-
-/**
- * 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);
-
-/**
- * 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);
-
-/**
- * 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
+ * Prints a listing containing all the filters supported by the
  * program.
  */
-int show_sample_fmts(void *optctx, const char *opt, const char *arg);
+void show_filters(void);
 
 /**
- * Return a positive value if a line read from standard input
- * starts with [yY], otherwise return 0.
+ * Prints a listing containing all the bit stream filters supported by the
+ * program.
+ */
+void show_bsfs(void);
+
+/**
+ * Prints a listing containing all the protocols supported by the
+ * program.
+ */
+void show_protocols(void);
+
+/**
+ * Prints a listing containing all the pixel formats supported by the
+ * program.
+ */
+void show_pix_fmts(void);
+
+/**
+ * Returns a positive value if reads from standard input a line
+ * starting with [yY], otherwise returns 0.
  */
 int read_yesno(void);
 
 /**
- * Read the file with name filename, and put its content in a newly
+ * Reads the file with name filename, and puts 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
+ * @param bufptr puts here the pointer to the newly allocated buffer
+ * @param size puts here the size of the newly allocated buffer
  * @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.
- *
- * If is_path is non-zero, look for the file in the path preset_name.
- * Otherwise search for a file named arg.ffpreset in the directories
- * $FFMPEG_DATADIR (if set), $HOME/.ffmpeg, and in the datadir defined
- * 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.
- *
- * @param filename buffer where the name of the found filename is written
- * @param filename_size size in bytes of the filename buffer
- * @param preset_name name of the preset to search
- * @param is_path tell if preset_name is a filename path
- * @param codec_name name of the codec for which to look for the
- * preset, may be NULL
- */
-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 */

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 | OPT_FUNC2, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },

common.mak

@@ -6,70 +6,60 @@
 all: all-yes
 
 ifndef SUBDIR
+vpath %.c   $(SRC_DIR)
+vpath %.h   $(SRC_DIR)
+vpath %.S   $(SRC_DIR)
+vpath %.asm $(SRC_DIR)
+vpath %.v   $(SRC_DIR)
+
+ifeq ($(SRC_DIR),$(SRC_PATH_BARE))
+BUILD_ROOT_REL = .
+else
+BUILD_ROOT_REL = ..
+endif
 
 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), \
-    $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
-$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
+    $(eval $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
+$(foreach VAR,$(SILENT),$(eval $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
 
-# 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 = $(call COMPILE,AS)
+CPPFLAGS := -I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(CPPFLAGS)
+CFLAGS   += $(ECFLAGS)
 
 %.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) $<
+%.ho: %.h
+	$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
 
-%.h.c:
-	$(Q)echo '#include "$*.h"' >$@
+%$(EXESUF): %.c
 
 %.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
 
 %.c %.h: TAG = GEN
 
-# Dummy rule to stop make trying to rebuild removed or renamed headers
-%.h:
-	@:
+install: install-libs install-headers
+install-libs: install-libs-yes
+
+uninstall: uninstall-libs uninstall-headers
+
+.PHONY: all depend dep *clean install* uninstall* examples testprogs
 
 # Disable suffix rules.  Most of the builtin rules are suffix rules,
 # so this saves some time on slow systems.
@@ -79,62 +69,41 @@ COMPILE_S = $(call COMPILE,AS)
 $(OBJS):
 endif
 
-include $(SRC_PATH)/arch.mak
+OBJS-$(HAVE_MMX) +=  $(MMX-OBJS-yes)
 
+CFLAGS    += $(CFLAGS-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   := $(addprefix -L$(BUILD_ROOT)/lib,$(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 := $(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))
+DEP_LIBS := $(foreach NAME,$(FFLIBS),$(BUILD_ROOT_REL)/lib$(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
+DEPS := $(OBJS:.o=.d)
+depend dep: $(DEPS)
 
-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 $(DEPS))

compat/getopt.c

@@ -1,86 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * This file was copied from the following newsgroup posting:
- *
- * Newsgroups: mod.std.unix
- * Subject: public domain AT&T getopt source
- * Date: 3 Nov 85 19:34:15 GMT
- *
- * Here's something you've all been waiting for:  the AT&T public domain
- * source for getopt(3).  It is the code which was given out at the 1985
- * UNIFORUM conference in Dallas.  I obtained it by electronic mail
- * directly from AT&T.  The people there assure me that it is indeed
- * in the public domain.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-static int opterr = 1;
-static int optind = 1;
-static int optopt;
-static char *optarg;
-
-#undef fprintf
-
-static int getopt(int argc, char *argv[], char *opts)
-{
-    static int sp = 1;
-    int c;
-    char *cp;
-
-    if (sp == 1) {
-        if (optind >= argc ||
-            argv[optind][0] != '-' || argv[optind][1] == '\0')
-            return EOF;
-        else if (!strcmp(argv[optind], "--")) {
-            optind++;
-            return EOF;
-        }
-    }
-    optopt = c = argv[optind][sp];
-    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
-        fprintf(stderr, ": illegal option -- %c\n", c);
-        if (argv[optind][++sp] == '\0') {
-            optind++;
-            sp = 1;
-        }
-        return '?';
-    }
-    if (*++cp == ':') {
-        if (argv[optind][sp+1] != '\0')
-            optarg = &argv[optind++][sp+1];
-        else if(++optind >= argc) {
-            fprintf(stderr, ": option requires an argument -- %c\n", c);
-            sp = 1;
-            return '?';
-        } else
-            optarg = argv[optind++];
-        sp = 1;
-    } else {
-        if (argv[optind][++sp] == '\0') {
-            sp = 1;
-            optind++;
-        }
-        optarg = NULL;
-    }
-
-    return c;
-}

compat/msvcrt/snprintf.c

@@ -1,71 +0,0 @@
-/*
- * C99-compatible snprintf() and vsnprintf() implementations
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <stdio.h>
-#include <stdarg.h>
-#include <limits.h>
-#include <string.h>
-
-#include "compat/va_copy.h"
-#include "libavutil/error.h"
-
-#if defined(__MINGW32__)
-#define EOVERFLOW EFBIG
-#endif
-
-int avpriv_snprintf(char *s, size_t n, const char *fmt, ...)
-{
-    va_list ap;
-    int ret;
-
-    va_start(ap, fmt);
-    ret = avpriv_vsnprintf(s, n, fmt, ap);
-    va_end(ap);
-
-    return ret;
-}
-
-int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
-                     va_list ap)
-{
-    int ret;
-    va_list ap_copy;
-
-    if (n == 0)
-        return _vscprintf(fmt, ap);
-    else if (n > INT_MAX)
-        return AVERROR(EOVERFLOW);
-
-    /* we use n - 1 here because if the buffer is not big enough, the MS
-     * runtime libraries don't add a terminating zero at the end. MSDN
-     * recommends to provide _snprintf/_vsnprintf() a buffer size that
-     * is one less than the actual buffer, and zero it before calling
-     * _snprintf/_vsnprintf() to workaround this problem.
-     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
-    memset(s, 0, n);
-    va_copy(ap_copy, ap);
-    ret = _vsnprintf(s, n - 1, fmt, ap_copy);
-    va_end(ap_copy);
-    if (ret == -1)
-        ret = _vscprintf(fmt, ap);
-
-    return ret;
-}

compat/plan9/head

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

compat/plan9/main.c

@@ -1,34 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-int plan9_main(int argc, char **argv);
-
-#undef main
-int main(int argc, char **argv)
-{
-    /* The setfcr() function in lib9 is broken, must use asm. */
-#ifdef __i386
-    short fcr;
-    __asm__ volatile ("fstcw        %0 \n"
-                      "or      $63, %0 \n"
-                      "fldcw        %0 \n"
-                      : "=m"(fcr));
-#endif
-
-    return plan9_main(argc, argv);
-}

compat/plan9/printf

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

compat/strtod.c

@@ -1,94 +0,0 @@
-/*
- * C99-compatible strtod() implementation
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <ctype.h>
-#include <limits.h>
-#include <stdlib.h>
-
-#include "libavutil/avstring.h"
-#include "libavutil/mathematics.h"
-
-static char *check_nan_suffix(char *s)
-{
-    char *start = s;
-
-    if (*s++ != '(')
-        return start;
-
-    while ((*s >= 'a' && *s <= 'z') || (*s >= 'A' && *s <= 'Z') ||
-           (*s >= '0' && *s <= '9') ||  *s == '_')
-        s++;
-
-    return *s == ')' ? s + 1 : start;
-}
-
-#undef strtod
-double strtod(const char *, char **);
-
-double avpriv_strtod(const char *nptr, char **endptr)
-{
-    char *end;
-    double res;
-
-    /* Skip leading spaces */
-    while (isspace(*nptr))
-        nptr++;
-
-    if (!av_strncasecmp(nptr, "infinity", 8)) {
-        end = nptr + 8;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "inf", 3)) {
-        end = nptr + 3;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+infinity", 9)) {
-        end = nptr + 9;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+inf", 4)) {
-        end = nptr + 4;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "-infinity", 9)) {
-        end = nptr + 9;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "-inf", 4)) {
-        end = nptr + 4;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "nan", 3)) {
-        end = check_nan_suffix(nptr + 3);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "+nan", 4) ||
-               !av_strncasecmp(nptr, "-nan", 4)) {
-        end = check_nan_suffix(nptr + 4);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "0x", 2) ||
-               !av_strncasecmp(nptr, "-0x", 3) ||
-               !av_strncasecmp(nptr, "+0x", 3)) {
-        /* FIXME this doesn't handle exponents, non-integers (float/double)
-         * and numbers too large for long long */
-        res = strtoll(nptr, &end, 16);
-    } else {
-        res = strtod(nptr, &end);
-    }
-
-    if (endptr)
-        *endptr = end;
-
-    return res;
-}

compat/tms470/math.h

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

compat/va_copy.h

@@ -1,26 +0,0 @@
-/*
- * MSVC Compatible va_copy macro
- * Copyright (c) 2012 Derek Buitenhuis
- *
- * 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 <stdarg.h>
-
-#if !defined(va_copy) && defined(_MSC_VER)
-#define va_copy(dst, src) ((dst) = (src))
-#endif

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

doc/RELEASE_NOTES

@@ -1,25 +0,0 @@
-Release Notes
-=============
-
-* 1.1 "Fire Flower"  January, 2013
-
-
-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.
-
-Bugreports against FFmpeg git master or the most recent FFmpeg release are
-accepted. If you are experiencing issues with any formally released version of
-FFmpeg, please try git master to check if the issue still exists. If it does,
-make your report against the development code following the usual bug reporting
-guidelines.
-
-Of big interest to our Windows users, FFmpeg now supports building with the MSVC
-compiler. Since MSVC does not support C99 features used extensively by FFmpeg,
-this has been accomplished using a converter that turns C99 code to C89. See the
-platform-specific documentation for more detailed documentation on building
-FFmpeg with MSVC.
-
-The used output sample format for several audio decoders has changed, make
-sure you always check/use AVCodecContext.sample_fmt or AVFrame.format.

doc/TODO

@@ -0,0 +1,90 @@
+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
+
+Francois' TODO list: (unordered, without any timeframe)
+-------------------
+- test MACE decoder against the openquicktime one as suggested by A'rpi
+- BeOS audio input grabbing backend
+- BeOS video input grabbing backend
+- publish my BeOS libposix on BeBits so I can officially support ffserver :)
+- check the whole code for thread-safety (global and init stuff)
+
+Philip'a TODO list: (alphabetically ordered) (please help)
+------------------
+- Add a multi-ffm filetype so that feeds can be recorded into multiple files rather
+  than one big file.
+- Authenticated users support -- where the authentication is in the URL
+- Change ASF files so that the embedded timestamp in the frames is right rather
+  than being an offset from the start of the stream
+- Make ffm files more resilient to changes in the codec structures so that you
+  can play old ffm files.
+
+Baptiste's TODO list:
+-----------------
+- mov edit list support (AVEditList)
+- YUV 10 bit per component support "2vuy"
+- mxf muxer
+- mpeg2 non linear quantizer
+
+unassigned TODO: (unordered)
+---------------
+- use AVFrame for audio codecs too
+- rework aviobuf.c buffering strategy and fix url_fskip
+- generate optimal huffman tables for mjpeg encoding
+- fix ffserver regression tests
+- support xvids motion estimation
+- support x264s motion estimation
+- support x264s rate control
+- SNOW: non translational motion compensation
+- SNOW: more optimal quantization
+- SNOW: 4x4 block support
+- SNOW: 1/8 pel motion compensation support
+- SNOW: iterative motion estimation based on subsampled images
+- SNOW: try B frames and MCTF and see how their PSNR/bitrate/complexity behaves
+- SNOW: try to use the wavelet transformed MC-ed reference frame as context for the entropy coder
+- SNOW: think about/analyize how to make snow use multiple cpus/threads
+- SNOW: finish spec
+- FLAC: lossy encoding (viterbi and naive scalar quantization)
+- libavfilter
+- JPEG2000 decoder & encoder
+- MPEG4 GMC encoding support
+- macroblock based pixel format (better cache locality, somewhat complex, one paper claimed it faster for high res)
+- regression tests for codecs which do not have an encoder (I+P-frame bitstream in svn)
+- add support for using mplayers video filters to ffmpeg
+- H264 encoder
+- per MB ratecontrol (so VCD and such do work better)
+- write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions
+- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc
+- generic audio mixing API
+- extract PES packetizer from PS muxer and use it for new TS muxer
+- implement automatic AVBistreamFilter activation
+- make cabac encoder use bytestream (see http://trac.videolan.org/x264/changeset/?format=diff&new=651)
+- merge imdct and windowing, the current code does considerable amounts of redundant work

doc/authors.texi

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

doc/avtools-common-opts.texi

@@ -1,211 +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.

doc/avutil.txt

@@ -19,6 +19,7 @@ integer.c               128bit integer math
 lls.c
 mathematics.c           greatest common divisor, integer sqrt, integer log2, ...
 mem.c                   memory allocation routines with guaranteed alignment
+softfloat.c
 
 Headers:
 bswap.h                 big/little/native-endian conversion code

doc/bitstream_filters.texi

@@ -1,91 +0,0 @@
-@chapter Bitstream Filters
-@c man begin BITSTREAM FILTERS
-
-When you configure your FFmpeg build, all the supported bitstream
-filters are enabled by default. You can list all available ones using
-the configure option @code{--list-bsfs}.
-
-You can disable all the bitstream filters using the configure option
-@code{--disable-bsfs}, and selectively enable any bitstream filter using
-the option @code{--enable-bsf=BSF}, or you can disable a particular
-bitstream filter using the option @code{--disable-bsf=BSF}.
-
-The option @code{-bsfs} of the ff* tools will display the list of
-all the supported bitstream filters included in your build.
-
-Below is a description of the currently available bitstream filters.
-
-@section aac_adtstoasc
-
-@section chomp
-
-@section dump_extradata
-
-@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
-
-Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
-
-MJPEG is a video codec wherein each video frame is essentially a
-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
-@end example
-
-Unfortunately, these chunks are incomplete JPEG images, because
-they lack the DHT segment required for decoding. Quoting from
-@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
-
-Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
-commented that "MJPEG, or at least the MJPEG in AVIs having the
-MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
-Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
-and it must use basic Huffman encoding, not arithmetic or
-progressive. . . . You can indeed extract the MJPEG frames and
-decode them with a regular JPEG decoder, but you have to prepend
-the DHT segment to them, or else the decoder won't have any idea
-how to decompress the data. The exact table necessary is given in
-the OpenDML spec."
-
-This bitstream filter patches the header of frames extracted from an MJPEG
-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
-exiftran -i -9 frame*.jpg
-ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
-@end example
-
-@section mjpega_dump_header
-
-@section movsub
-
-@section mp3_header_compress
-
-@section mp3_header_decompress
-
-@section noise
-
-@section remove_extradata
-
-@c man end BITSTREAM FILTERS

doc/build_system.txt

@@ -1,50 +0,0 @@
-FFmpeg currently uses a custom build system, this text attempts to document
-some of its obscure features and options.
-
-Makefile variables:
-
-V
-    Disable the default terse mode, the full command issued by make and its
-    output will be shown on the screen.
-
-DESTDIR
-    Destination directory for the install targets, useful to prepare packages
-    or install FFmpeg in cross-environments.
-
-Makefile targets:
-
-all
-    Default target, builds all the libraries and the executables.
-
-fate
-    Run the fate test suite, note you must have installed it
-
-fate-list
-    Will list all fate/regression test targets
-
-install
-    Install headers, libraries and programs.
-
-libavformat/output-example
-    Build the libavformat basic example.
-
-libavcodec/api-example
-    Build the libavcodec basic example.
-
-libswscale/swscale-test
-    Build the swscale self-test (useful also as example).
-
-
-Useful standard make commands:
-make -t <target>
-    Touch all files that otherwise would be build, this is useful to reduce
-    unneeded rebuilding when changing headers, but note you must force rebuilds
-    of files that actually need it by hand then.
-
-make -j<num>
-    rebuild with multiple jobs at the same time. Faster on multi processor systems
-
-make -k
-    continue build in case of errors, this is useful for the regression tests
-    sometimes but note it will still not run all reg tests.
-

doc/decoders.texi

@@ -1,161 +0,0 @@
-@chapter Decoders
-@c man begin DECODERS
-
-Decoders are configured elements in FFmpeg which allow the decoding of
-multimedia streams.
-
-When you configure your FFmpeg build, all the supported native decoders
-are enabled by default. Decoders requiring an external library must be enabled
-manually via the corresponding @code{--enable-lib} option. You can list all
-available decoders using the configure option @code{--list-decoders}.
-
-You can disable all the decoders with the configure option
-@code{--disable-decoders} and selectively enable / disable single decoders
-with the options @code{--enable-decoder=@var{DECODER}} /
-@code{--disable-decoder=@var{DECODER}}.
-
-The option @code{-codecs} of the ff* tools will display the list of
-enabled decoders.
-
-@c man end DECODERS
-
-@chapter Video Decoders
-@c man begin VIDEO DECODERS
-
-A description of some of the currently available video decoders
-follows.
-
-@section rawvideo
-
-Raw video decoder.
-
-This decoder decodes rawvideo streams.
-
-@subsection Options
-
-@table @option
-@item top @var{top_field_first}
-Specify the assumed field type of the input video.
-@table @option
-@item -1
-the video is assumed to be progressive (default)
-@item 0
-bottom-field-first is assumed
-@item 1
-top-field-first is assumed
-@end table
-
-@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

doc/default.css

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

doc/demuxers.texi

@@ -1,239 +0,0 @@
-@chapter Demuxers
-@c man begin DEMUXERS
-
-Demuxers are configured elements in FFmpeg which allow to read the
-multimedia streams from a particular type of file.
-
-When you configure your FFmpeg build, all the supported demuxers
-are enabled by default. You can list all available ones using the
-configure option "--list-demuxers".
-
-You can disable all the demuxers using the configure option
-"--disable-demuxers", and selectively enable a single demuxer with
-the option "--enable-demuxer=@var{DEMUXER}", or disable it
-with the option "--disable-demuxer=@var{DEMUXER}".
-
-The option "-formats" of the ff* tools will display the list of
-enabled demuxers.
-
-The description of some of the currently available demuxers follows.
-
-@section image2
-
-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
-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
-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
-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.
-
-For example the pattern "img-%03d.bmp" will match a sequence of
-filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
-@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
-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.
-
-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
-@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.
-
-This demuxer presents all AVStreams from all variant streams.
-The id field is set to the bitrate variant index number. By setting
-the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
-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

doc/developer.texi

@@ -2,25 +2,23 @@
 
 @settitle Developer Documentation
 @titlepage
+@sp 7
 @center @titlefont{Developer Documentation}
+@sp 3
 @end titlepage
 
-@top
-
-@contents
 
 @chapter Developers Guide
 
 @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 +33,59 @@ 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'.
+rejected by the Subversion repository.
 
 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 +113,11 @@ 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
 
@@ -243,9 +149,9 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
    Also if you have doubts about splitting or not splitting, do not hesitate to
    ask/discuss it on the developer mailing list.
 @item
-   Do not change behavior of the programs (renaming options etc) or public
-   API or ABI without first discussing it on the ffmpeg-devel mailing list.
-   Do not remove functionality from the code. Just improve!
+   Do not change behavior of the program (renaming options etc) without
+   first discussing it on the ffmpeg-devel mailing list. Do not remove
+   functionality from the code. Just improve!
 
    Note: Redundant code can be removed.
 @item
@@ -273,13 +179,10 @@ 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:
-   area changed: Short 1 line description
-
-   details describing what and why and giving references.
 @item
-   Make sure the author of the commit is set correctly. (see git commit --author)
-   If you apply a patch, send an
+   If you apply a patch by someone else, include the name and email address in
+   the log message. Since the ffmpeg-cvslog mailing list is publicly
+   archived you should add some SPAM protection to the email address. Send an
    answer to ffmpeg-devel (or wherever you got the patch from) saying that
    you applied the patch.
 @item
@@ -308,7 +211,7 @@ 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 +220,7 @@ 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
@@ -336,14 +238,12 @@ 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, (@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 :-)
+When you submit your patch, try to send a unified diff (diff '-up'
+option). We cannot read other diffs :-)
 
 Also please do not submit a patch which contains several unrelated changes.
 Split it into separate, self-contained pieces. This does not mean splitting
@@ -355,13 +255,13 @@ 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
 transmission) to the ffmpeg-devel mailing list, see
-@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
+@url{http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel}
 
 It also helps quite a bit if you tell us what the patch does (for example
 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
@@ -389,16 +289,14 @@ send a reminder by email. Your patch should eventually be dealt with.
     Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
     AVInputFormat/AVOutputFormat struct?
 @item
-    Did you bump the minor version number (and reset the micro version
-    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
+    Did you bump the minor version number in @file{avcodec.h} or
+    @file{avformat.h}?
 @item
     Did you register it in @file{allcodecs.c} or @file{allformats.c}?
 @item
-    Did you add the 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?
@@ -413,31 +311,22 @@ send a reminder by email. Your patch should eventually be dealt with.
     If it depends on a parser or a library, did you add that dependency in
     configure?
 @item
-    Did you @code{git add} the appropriate files before committing?
-@item
-    Did you make sure it compiles standalone, i.e. with
-    @code{configure --disable-everything --enable-decoder=foo}
-    (or @code{--enable-demuxer} or whatever your component is)?
+    Did you "svn add" the appropriate files before commiting?
 @end enumerate
 
-
 @section patch submission checklist
 
 @enumerate
 @item
-    Does @code{make fate} pass with the patch applied?
+    Do the regression tests pass with the patch applied?
 @item
-    Was the patch generated with git format-patch or send-email?
+    Does @code{make checkheaders} pass with the patch applied?
 @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
-    of sign off.
+    Is the patch a unified diff?
 @item
-    Did you provide a clear git commit log message?
+    Is the patch against latest FFmpeg SVN?
 @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
@@ -453,6 +342,9 @@ send a reminder by email. Your patch should eventually be dealt with.
     Did you test your decoder or demuxer against damaged data? If no, see
     tools/trasher and the noise bitstream filter. Your decoder or demuxer
     should not crash or end in a (near) infinite loop when fed damaged data.
+@item
+    Is the patch created from the root of the source tree, so it can be
+    applied with @code{patch -p0}?
 @item
     Does the patch not mix functional and cosmetic changes?
 @item
@@ -489,19 +381,13 @@ send a reminder by email. Your patch should eventually be dealt with.
     Lines with similar content should be aligned vertically when doing so
     improves readability.
 @item
-    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.
+    Did you provide a suggestion for a clear commit log message?
 @end enumerate
 
 @section Patch review process
 
 All patches posted to ffmpeg-devel will be reviewed, unless they contain a
-clear note that the patch is not for the git master branch.
+clear note that the patch is not for SVN.
 Reviews and comments will be posted as replies to the patch on the
 mailing list. The patch submitter then has to take care of every comment,
 that can be by resubmitting a changed patch or by discussion. Resubmitted
@@ -514,38 +400,37 @@ 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

doc/doxy-wrapper.sh

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

doc/doxy/footer.html

@@ -1,9 +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>

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">

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

doc/eval.texi

@@ -1,252 +0,0 @@
-@chapter Expression Evaluation
-@c man begin EXPRESSION EVALUATION
-
-When evaluating an arithmetic expression, FFmpeg uses an internal
-formula evaluator, implemented through the @file{libavutil/eval.h}
-interface.
-
-An expression may contain unary, binary operators, constants, and
-functions.
-
-Two expressions @var{expr1} and @var{expr2} can be combined to form
-another expression "@var{expr1};@var{expr2}".
-@var{expr1} and @var{expr2} are evaluated in turn, and the new
-expression evaluates to the value of @var{expr2}.
-
-The following binary operators are available: @code{+}, @code{-},
-@code{*}, @code{/}, @code{^}.
-
-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}).
-The function returns the loaded value.
-
-@item while(cond, expr)
-Evaluate expression @var{expr} while the expression @var{cond} is
-non-zero, and returns the value of the last @var{expr} evaluation, or
-NAN if @var{cond} was always false.
-
-@item ceil(expr)
-Round the value of expression @var{expr} upwards to the nearest
-integer. For example, "ceil(1.5)" is "2.0".
-
-@item floor(expr)
-Round the value of expression @var{expr} downwards to the nearest
-integer. For example, "floor(-1.5)" is "-2.0".
-
-@item trunc(expr)
-Round the value of expression @var{expr} towards zero to the nearest
-integer. For example, "trunc(-1.5)" is "-1.0".
-
-@item sqrt(expr)
-Compute the square root of @var{expr}. This is equivalent to
-"(@var{expr})^.5".
-
-@item not(expr)
-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:
-
-@code{*} works like AND
-
-@code{+} works like OR
-
-and the construct:
-@example
-if A then B else C
-@end example
-is equivalent to
-@example
-if(A,B) + ifnot(A,C)
-@end example
-
-In your C code, you can extend the list of unary and binary functions,
-and define recognized constants, so that they are available for your
-expressions.
-
-The evaluator also recognizes the International System number
-postfixes. 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.
-
-Follows the list of available International System postfixes, with
-indication of the corresponding powers of 10 and of 2.
-@table @option
-@item y
--24 / -80
-@item z
--21 / -70
-@item a
--18 / -60
-@item f
--15 / -50
-@item p
--12 / -40
-@item n
--9 / -30
-@item u
--6 / -20
-@item m
--3 / -10
-@item c
--2
-@item d
--1
-@item h
-2
-@item k
-3 / 10
-@item K
-3 / 10
-@item M
-6 / 20
-@item G
-9 / 30
-@item T
-12 / 40
-@item P
-15 / 40
-@item E
-18 / 50
-@item Z
-21 / 60
-@item Y
-24 / 70
-@end table
-
-@c man end

doc/examples/Makefile

@@ -1,37 +0,0 @@
-# use pkg-config for getting CFLAGS and LDLIBS
-FFMPEG_LIBS=    libavdevice                        \
-                libavformat                        \
-                libavfilter                        \
-                libavcodec                         \
-                libswresample                      \
-                libswscale                         \
-                libavutil                          \
-
-CFLAGS += -Wall -O2 -g
-CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
-LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
-
-EXAMPLES=       decoding_encoding                  \
-                demuxing                           \
-                filtering_video                    \
-                filtering_audio                    \
-                metadata                           \
-                muxing                             \
-                resampling_audio                   \
-                scaling_video                      \
-
-OBJS=$(addsuffix .o,$(EXAMPLES))
-
-# the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
-muxing:            LDLIBS += -lm
-
-.phony: all clean-test clean
-
-all: $(OBJS) $(EXAMPLES)
-
-clean-test:
-	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
-
-clean: clean-test
-	$(RM) $(EXAMPLES) $(OBJS)

doc/examples/README

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

doc/examples/decoding_encoding.c

@@ -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;
-}

doc/examples/demuxing.c

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

doc/examples/filtering_audio.c

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

doc/examples/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);
-}

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;
-}

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;
-}

doc/examples/resampling_audio.c

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

doc/examples/scaling_video.c

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

doc/faq.texi

@@ -2,15 +2,29 @@
 
 @settitle FFmpeg FAQ
 @titlepage
+@sp 7
 @center @titlefont{FFmpeg FAQ}
+@sp 3
 @end titlepage
 
-@top
-
-@contents
 
 @chapter General Questions
 
+@section When will the next FFmpeg version be released? / Why are FFmpeg releases so few and far between?
+
+Like most open source projects FFmpeg suffers from a certain lack of
+manpower. For this reason the developers have to prioritize the work
+they do and putting out releases is not at the top of the list, fixing
+bugs and reviewing patches takes precedence. Please don't complain or
+request more timely and/or frequent releases unless you are willing to
+help out creating them.
+
+@section I have a problem with an old version of FFmpeg; where should I report it?
+Nowhere. Upgrade to the latest release or if there is no recent release upgrade
+to Subversion HEAD. You could also try to report it. Maybe you will get lucky and
+become the first person in history to get an answer different from "upgrade
+to Subversion HEAD".
+
 @section Why doesn't FFmpeg support feature [xyz]?
 
 Because no one has taken on that task yet. FFmpeg development is
@@ -24,6 +38,30 @@ 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 bugreports.html.
+@item You didn't use Subversion HEAD.
+@item You reported a segmentation fault without gdb output.
+@item You describe a problem but not how to reproduce it.
+@item It's unclear if you use ffmpeg as command line tool or use
+libav* from another application.
+@item You speak about a video having problems on playback but
+not what you use to play it.
+@item We have no faint clue what you are talking about besides
+that it is related to FFmpeg.
+@end itemize
+
+@section Is there a forum for FFmpeg? I do not like mailing lists.
+
+You may view our mailing lists with a more forum-alike look here:
+@url{http://dir.gmane.org/gmane.comp.video.ffmpeg.user},
+but, if you post, please remember that our mailing list rules still apply there.
+
 @section I cannot read this file although this format seems to be supported by ffmpeg.
 
 Even if ffmpeg can read the container format, it may not support all its
@@ -47,7 +85,7 @@ The following list of video codecs should work on most Windows systems:
 .asf only
 @item mpeg4
 Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
-@item mpeg1video
+@item mpeg1
 .mpg only
 @end table
 Note, ASF files often have .wmv or .wma extensions in Windows. It should also
@@ -59,9 +97,9 @@ The following list of audio codecs should work on most Windows systems:
 @table @option
 @item adpcm_ima_wav
 @item adpcm_ms
-@item pcm_s16le
+@item pcm
 always
-@item libmp3lame
+@item mp3
 If some MP3 codec like LAME is installed.
 @end table
 
@@ -79,23 +117,11 @@ 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 +136,7 @@ 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
@@ -128,7 +145,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
 @file{img001.jpg}, @file{img002.jpg} and so on.
 
 @example
-  x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+  x=1; for i in *jpg; do counter=$(printf %03d $x); ln "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
 @end example
 
 If you want to sequence them by oldest modified first, substitute
@@ -142,12 +159,6 @@ 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 +172,15 @@ 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.
@@ -184,28 +195,79 @@ quite logical that there is a small reduction of quality. This is not a bug.
 
 Use @file{-} as file name.
 
+@section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux?
+
+This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at
+(@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-420-2.4.26.patch?view=co}). This may also
+apply cleanly to other 2.4-series kernels.
+
+@section How do I avoid the ugly aliasing artifacts in bt8x8 captures on Linux?
+
+Pass 'combfilter=1 lumafilter=1' to the bttv driver. Note though that 'combfilter=1'
+will cause somewhat too strong filtering. A fix is to apply (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.4.26.patch?view=co})
+or (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.6.6.patch?view=co})
+and pass 'combfilter=2'.
+
 @section -f jpeg doesn't work.
 
 Try '-f image2 test%d.jpg'.
 
-@section Why can I not change the 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
+4mv, title
+@item non-working stuff
+B-frames
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X output.mp4
+@end table
+
+@section How do I encode videos which play on the PSP?
+
+@table @option
+@item needed stuff
+-acodec libfaac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp
+@item working stuff
+4mv, title
+@item non-working stuff
+B-frames
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp output.mp4
+@item needed stuff for H.264
+-acodec libfaac -vcodec libx264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
+@item working stuff for H.264
+title, loop filter
+@item non-working stuff for H.264
+CAVLC
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
+@item higher resolution for newer PSP firmwares, width<=480, height<=272
+-vcodec libx264 -level 21 -coder 1 -f psp
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -ac 2 -ar 48000 -vcodec libx264 -level 21 -b 640kb -coder 1 -f psp -flags +loop -trellis 2 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -s 480x272 output.mp4
+@end table
+
 @section Which are good parameters for encoding high quality MPEG-4?
 
-'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
+'-mbd rd -flags +4mv+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
 things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
 
 @section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
@@ -224,78 +286,24 @@ material, and try '-top 0/1' if the result looks really messed-up.
 If you have built FFmpeg with @code{./configure --enable-avisynth}
 (only possible on MinGW/Cygwin platforms),
 then you may use any file that DirectShow can read as input.
+(Be aware that this feature has been recently added,
+so you will need to help yourself in case of problems.)
 
 Just create an "input.avs" text file with this single line ...
 @example
   DirectShowSource("C:\path to your file\yourfile.asf")
 @end example
-... and then feed that text file to ffmpeg:
+... and then feed that text file to FFmpeg:
 @example
   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 +311,28 @@ 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 +340,7 @@ 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 +357,33 @@ 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 FFmpeg does not adhere to 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}).
 
@@ -455,51 +395,64 @@ 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?
 
 The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
 information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
-you need the debug information, use the *_g versions.
+you need the debug information, used the *_g versions.
 
 @section I do not like the LGPL, can I contribute code under the GPL instead?
 
 Yes, as long as the code is optional and can easily and cleanly be placed
-under #if CONFIG_GPL without breaking anything. So, for example, a new codec
+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 +460,23 @@ 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 libavformat/file.c in FFmpeg
+and libmpdemux/demux_lavf.c in MPlayer sources.
+
+@section I get "No compatible shell script interpreter found." in MSys.
+
+The standard MSys bash (2.04) is broken. You need to install 2.05 or later.
+
+@section I get "./configure: line <xxx>: pr: command not found" in MSys.
+
+The standard MSys install doesn't come with pr. You need to get it from the coreutils package.
+
+@section I tried to pass RTP packets into a decoder, but it doesn't work.
+
+RTP is a container format like any other, you must first depacketize the
+codec frames/samples stored in RTP and then feed to the decoder.
 
 @section Where can I find libav* headers for Pascal/Delphi?
 
@@ -523,7 +484,7 @@ see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
 
 @section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
 
-see @url{http://www.ffmpeg.org/~michael/}
+see @url{http://svn.ffmpeg.org/michael/trunk/docs/}
 
 @section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
 
@@ -531,24 +492,12 @@ 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

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

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'

doc/ffmpeg-bitstream-filters.texi

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

doc/ffmpeg-devices.texi

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

doc/ffmpeg-doc.texi

@@ -0,0 +1,956 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFmpeg Documentation
+@titlepage
+@sp 7
+@center @titlefont{FFmpeg Documentation}
+@sp 3
+@end titlepage
+
+
+@chapter Introduction
+
+FFmpeg is a very fast video and audio converter. It can also grab from
+a live audio/video source.
+
+The command line interface is designed to be intuitive, in the sense
+that FFmpeg tries to figure out all parameters that can possibly be
+derived automatically. You usually only have to specify the target
+bitrate you want.
+
+FFmpeg can also convert from any sample rate to any other, and resize
+video on the fly with a high quality polyphase filter.
+
+@chapter Quick Start
+
+@c man begin EXAMPLES
+@section Video and Audio grabbing
+
+FFmpeg can grab video and audio from devices given that you specify the input
+format and device.
+
+@example
+ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
+@end example
+
+Note that you must activate the right video source and channel before
+launching FFmpeg with any TV viewer such as xawtv
+(@url{http://linux.bytesex.org/xawtv/}) by Gerd Knorr. You also
+have to set the audio recording levels correctly with a
+standard mixer.
+
+@section X11 grabbing
+
+FFmpeg can grab the X11 display.
+
+@example
+ffmpeg -f x11grab -s cif -i :0.0 /tmp/out.mpg
+@end example
+
+0.0 is display.screen number of your X11 server, same as
+the DISPLAY environment variable.
+
+@example
+ffmpeg -f x11grab -s cif -i :0.0+10,20 /tmp/out.mpg
+@end example
+
+0.0 is display.screen number of your X11 server, same as the DISPLAY environment
+variable. 10 is the x-offset and 20 the y-offset for the grabbing.
+
+@section Video and Audio file format conversion
+
+* FFmpeg can use any supported file format and protocol as input:
+
+Examples:
+
+* You can use YUV files as input:
+
+@example
+ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
+@end example
+
+It will use the files:
+@example
+/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
+/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
+@end example
+
+The Y files use twice the resolution of the U and V files. They are
+raw files, without header. They can be generated by all decent video
+decoders. You must specify the size of the image with the @option{-s} option
+if FFmpeg cannot guess it.
+
+* You can input from a raw YUV420P file:
+
+@example
+ffmpeg -i /tmp/test.yuv /tmp/out.avi
+@end example
+
+test.yuv is a file containing raw YUV planar data. Each frame is composed
+of the Y plane followed by the U and V planes at half vertical and
+horizontal resolution.
+
+* You can output to a raw YUV420P file:
+
+@example
+ffmpeg -i mydivx.avi hugefile.yuv
+@end example
+
+* You can set several input files and output files:
+
+@example
+ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
+@end example
+
+Converts the audio file a.wav and the raw YUV video file a.yuv
+to MPEG file a.mpg.
+
+* You can also do audio and video conversions at the same time:
+
+@example
+ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
+@end example
+
+Converts a.wav to MPEG audio at 22050 Hz sample rate.
+
+* You can encode to several formats at the same time and define a
+mapping from input stream to output streams:
+
+@example
+ffmpeg -i /tmp/a.wav -ab 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
+@end example
+
+Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map
+file:index' specifies which input stream is used for each output
+stream, in the order of the definition of output streams.
+
+* You can transcode decrypted VOBs:
+
+@example
+ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
+@end example
+
+This is a typical DVD ripping example; the input is a VOB file, the
+output an AVI file with MPEG-4 video and MP3 audio. Note that in this
+command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
+GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
+input video. Furthermore, the audio stream is MP3-encoded so you need
+to enable LAME support by passing @code{--enable-libmp3lame} to configure.
+The mapping is particularly useful for DVD transcoding
+to get the desired audio language.
+
+NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
+
+* You can extract images from a video, or create a video from many images:
+
+For extracting images from a video:
+@example
+ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
+@end example
+
+This will extract one video frame per second from the video and will
+output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
+etc. Images will be rescaled to fit the new WxH values.
+
+If you want to extract just a limited number of frames, you can use the
+above command in combination with the -vframes or -t option, or in
+combination with -ss to start extracting from a certain point in time.
+
+For creating a video from many images:
+@example
+ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
+@end example
+
+The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
+composed of three digits padded with zeroes to express the sequence
+number. It is the same syntax supported by the C printf function, but
+only formats accepting a normal integer are suitable.
+
+* You can put many streams of the same type in the output:
+
+@example
+ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
+@end example
+
+In addition to the first video and audio streams, the resulting
+output file @file{test12.avi} will contain the second video
+and the second audio stream found in the input streams list.
+
+The @code{-newvideo}, @code{-newaudio} and @code{-newsubtitle}
+options have to be specified immediately after the name of the output
+file to which you want to add them.
+@c man end
+
+@chapter Invocation
+
+@section Syntax
+
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
+@c man end
+@end example
+@c man begin DESCRIPTION
+As a general rule, options are applied to the next specified
+file. Therefore, order is important, and you can have the same
+option on the command line multiple times. Each occurrence is
+then applied to the next input or output file.
+
+* To set the video bitrate of the output file to 64kbit/s:
+@example
+ffmpeg -i input.avi -b 64k output.avi
+@end example
+
+* To force the frame rate of the output file to 24 fps:
+@example
+ffmpeg -i input.avi -r 24 output.avi
+@end example
+
+* To force the frame rate of the input file (valid for raw formats only)
+to 1 fps and the frame rate of the output file to 24 fps:
+@example
+ffmpeg -r 1 -i input.m2v -r 24 output.avi
+@end example
+
+The format option may be needed for raw input files.
+
+By default, FFmpeg tries to convert as losslessly as possible: It
+uses the same audio and video parameters for the outputs as the one
+specified for the inputs.
+@c man end
+
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+
+@item -f @var{fmt}
+Force format.
+
+@item -i @var{filename}
+input file name
+
+@item -y
+Overwrite output files.
+
+@item -t @var{duration}
+Restrict the transcoded/captured video sequence
+to the duration specified in seconds.
+@code{hh:mm:ss[.xxx]} syntax is also supported.
+
+@item -fs @var{limit_size}
+Set the file size limit.
+
+@item -ss @var{position}
+Seek to given time position in seconds.
+@code{hh:mm:ss[.xxx]} syntax is also supported.
+
+@item -itsoffset @var{offset}
+Set the input time offset in seconds.
+@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
+This option affects all the input files that follow it.
+The offset is added to the timestamps of the input files.
+Specifying a positive offset means that the corresponding
+streams are delayed by 'offset' seconds.
+
+@item -timestamp @var{time}
+Set the timestamp.
+
+@item -metadata @var{key}=@var{value}
+Set a metadata key/value pair.
+
+For example, for setting the title in the output file:
+@example
+ffmpeg -i in.avi -metadata title="my title" out.flv
+@end example
+
+@item -v @var{number}
+Set the logging verbosity level.
+
+@item -target @var{type}
+Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd",
+"ntsc-svcd", ... ). All the format options (bitrate, codecs,
+buffer sizes) are then set automatically. You can just type:
+
+@example
+ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
+@end example
+
+Nevertheless you can specify additional options as long as you know
+they do not conflict with the standard, as in:
+
+@example
+ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
+@end example
+
+@item -dframes @var{number}
+Set the number of data frames to record.
+
+@item -scodec @var{codec}
+Force subtitle codec ('copy' to copy stream).
+
+@item -newsubtitle
+Add a new subtitle stream to the current output stream.
+
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
+
+@end table
+
+@section Video Options
+
+@table @option
+@item -b @var{bitrate}
+Set the video bitrate in bit/s (default = 200 kb/s).
+@item -vframes @var{number}
+Set the number of video frames to record.
+@item -r @var{fps}
+Set frame rate (Hz value, fraction or abbreviation), (default = 25).
+@item -s @var{size}
+Set frame size. The format is @samp{wxh} (ffserver default = 160x128, ffmpeg default = same as source).
+The following abbreviations are recognized:
+@table @samp
+@item sqcif
+128x96
+@item qcif
+176x144
+@item cif
+352x288
+@item 4cif
+704x576
+@item 16cif
+1408x1152
+@item qqvga
+160x120
+@item qvga
+320x240
+@item vga
+640x480
+@item svga
+800x600
+@item xga
+1024x768
+@item uxga
+1600x1200
+@item qxga
+2048x1536
+@item sxga
+1280x1024
+@item qsxga
+2560x2048
+@item hsxga
+5120x4096
+@item wvga
+852x480
+@item wxga
+1366x768
+@item wsxga
+1600x1024
+@item wuxga
+1920x1200
+@item woxga
+2560x1600
+@item wqsxga
+3200x2048
+@item wquxga
+3840x2400
+@item whsxga
+6400x4096
+@item whuxga
+7680x4800
+@item cga
+320x200
+@item ega
+640x350
+@item hd480
+852x480
+@item hd720
+1280x720
+@item hd1080
+1920x1080
+@end table
+
+@item -aspect @var{aspect}
+Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777).
+@item -croptop @var{size}
+Set top crop band size (in pixels).
+@item -cropbottom @var{size}
+Set bottom crop band size (in pixels).
+@item -cropleft @var{size}
+Set left crop band size (in pixels).
+@item -cropright @var{size}
+Set right crop band size (in pixels).
+@item -padtop @var{size}
+Set top pad band size (in pixels).
+@item -padbottom @var{size}
+Set bottom pad band size (in pixels).
+@item -padleft @var{size}
+Set left pad band size (in pixels).
+@item -padright @var{size}
+Set right pad band size (in pixels).
+@item -padcolor @var{hex_color}
+Set color of padded bands. The value for padcolor is expressed
+as a six digit hexadecimal number where the first two digits
+represent red, the middle two digits green and last two digits
+blue (default = 000000 (black)).
+@item -vn
+Disable video recording.
+@item -bt @var{tolerance}
+Set video bitrate tolerance (in bits, default 4000k).
+Has a minimum value of: (target_bitrate/target_framerate).
+In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
+willing to deviate from the target average bitrate value. This is
+not related to min/max bitrate. Lowering tolerance too much has
+an adverse effect on quality.
+@item -maxrate @var{bitrate}
+Set max video bitrate (in bit/s).
+Requires -bufsize to be set.
+@item -minrate @var{bitrate}
+Set min video bitrate (in bit/s).
+Most useful in setting up a CBR encode:
+@example
+ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
+@end example
+It is of little use elsewise.
+@item -bufsize @var{size}
+Set video buffer verifier buffer size (in bits).
+@item -vcodec @var{codec}
+Force video codec to @var{codec}. Use the @code{copy} special value to
+tell that the raw codec data must be copied as is.
+@item -sameq
+Use same video quality as source (implies VBR).
+
+@item -pass @var{n}
+Select the pass number (1 or 2). It is used to do two-pass
+video encoding. The statistics of the video are recorded in the first
+pass into a log file (see also the option -passlogfile),
+and in the second pass that log file is used to generate the video
+at the exact requested bitrate.
+On pass 1, you may just deactivate audio and set output to null,
+examples for Windows and Unix:
+@example
+ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL
+ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
+@end example
+
+@item -passlogfile @var{prefix}
+Set two-pass log file name prefix to @var{prefix}, the default file name
+prefix is ``ffmpeg2pass''. The complete file name will be
+@file{PREFIX-N.log}, where N is a number specific to the output
+stream.
+
+@item -newvideo
+Add a new video stream to the current output stream.
+
+@item -vlang @var{code}
+Set the ISO 639 language code (3 letters) of the current video stream.
+
+@end table
+
+@section Advanced Video Options
+
+@table @option
+@item -pix_fmt @var{format}
+Set pixel format. Use 'list' as parameter to show all the supported
+pixel formats.
+@item -sws_flags @var{flags}
+Set SwScaler flags.
+@item -g @var{gop_size}
+Set the group of pictures size.
+@item -intra
+Use only intra frames.
+@item -vdt @var{n}
+Discard threshold.
+@item -qscale @var{q}
+Use fixed video quantizer scale (VBR).
+@item -qmin @var{q}
+minimum video quantizer scale (VBR)
+@item -qmax @var{q}
+maximum video quantizer scale (VBR)
+@item -qdiff @var{q}
+maximum difference between the quantizer scales (VBR)
+@item -qblur @var{blur}
+video quantizer scale blur (VBR) (range 0.0 - 1.0)
+@item -qcomp @var{compression}
+video quantizer scale compression (VBR) (default 0.5).
+Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
+
+@item -lmin @var{lambda}
+minimum video lagrange factor (VBR)
+@item -lmax @var{lambda}
+max video lagrange factor (VBR)
+@item -mblmin @var{lambda}
+minimum macroblock quantizer scale (VBR)
+@item -mblmax @var{lambda}
+maximum macroblock quantizer scale (VBR)
+
+These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
+but you may use the QP2LAMBDA constant to easily convert from 'q' units:
+@example
+ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
+@end example
+
+@item -rc_init_cplx @var{complexity}
+initial complexity for single pass encoding
+@item -b_qfactor @var{factor}
+qp factor between P- and B-frames
+@item -i_qfactor @var{factor}
+qp factor between P- and I-frames
+@item -b_qoffset @var{offset}
+qp offset between P- and B-frames
+@item -i_qoffset @var{offset}
+qp offset between P- and I-frames
+@item -rc_eq @var{equation}
+Set rate control equation (@pxref{FFmpeg formula
+evaluator}) (default = @code{tex^qComp}).
+@item -rc_override @var{override}
+rate control override for specific intervals
+@item -me_method @var{method}
+Set motion estimation method to @var{method}.
+Available methods are (from lowest to best quality):
+@table @samp
+@item zero
+Try just the (0, 0) vector.
+@item phods
+@item log
+@item x1
+@item hex
+@item umh
+@item epzs
+(default method)
+@item full
+exhaustive search (slow and marginally better than epzs)
+@end table
+
+@item -dct_algo @var{algo}
+Set DCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_DCT_AUTO (default)
+@item 1
+FF_DCT_FASTINT
+@item 2
+FF_DCT_INT
+@item 3
+FF_DCT_MMX
+@item 4
+FF_DCT_MLIB
+@item 5
+FF_DCT_ALTIVEC
+@end table
+
+@item -idct_algo @var{algo}
+Set IDCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_IDCT_AUTO (default)
+@item 1
+FF_IDCT_INT
+@item 2
+FF_IDCT_SIMPLE
+@item 3
+FF_IDCT_SIMPLEMMX
+@item 4
+FF_IDCT_LIBMPEG2MMX
+@item 5
+FF_IDCT_PS2
+@item 6
+FF_IDCT_MLIB
+@item 7
+FF_IDCT_ARM
+@item 8
+FF_IDCT_ALTIVEC
+@item 9
+FF_IDCT_SH4
+@item 10
+FF_IDCT_SIMPLEARM
+@end table
+
+@item -er @var{n}
+Set error resilience to @var{n}.
+@table @samp
+@item 1
+FF_ER_CAREFUL (default)
+@item 2
+FF_ER_COMPLIANT
+@item 3
+FF_ER_AGGRESSIVE
+@item 4
+FF_ER_VERY_AGGRESSIVE
+@end table
+
+@item -ec @var{bit_mask}
+Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
+the following values:
+@table @samp
+@item 1
+FF_EC_GUESS_MVS (default = enabled)
+@item 2
+FF_EC_DEBLOCK (default = enabled)
+@end table
+
+@item -bf @var{frames}
+Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
+@item -mbd @var{mode}
+macroblock decision
+@table @samp
+@item 0
+FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).
+@item 1
+FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
+@item 2
+FF_MB_DECISION_RD: rate distortion
+@end table
+
+@item -4mv
+Use four motion vector by macroblock (MPEG-4 only).
+@item -part
+Use data partitioning (MPEG-4 only).
+@item -bug @var{param}
+Work around encoder bugs that are not auto-detected.
+@item -strict @var{strictness}
+How strictly to follow the standards.
+@item -aic
+Enable Advanced intra coding (h263+).
+@item -umv
+Enable Unlimited Motion Vector (h263+)
+
+@item -deinterlace
+Deinterlace pictures.
+@item -ilme
+Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
+Use this option if your input file is interlaced and you want
+to keep the interlaced format for minimum losses.
+The alternative is to deinterlace the input stream with
+@option{-deinterlace}, but deinterlacing introduces losses.
+@item -psnr
+Calculate PSNR of compressed frames.
+@item -vstats
+Dump video coding statistics to @file{vstats_HHMMSS.log}.
+@item -vstats_file @var{file}
+Dump video coding statistics to @var{file}.
+@item -top @var{n}
+top=1/bottom=0/auto=-1 field first
+@item -dc @var{precision}
+Intra_dc_precision.
+@item -vtag @var{fourcc/tag}
+Force video tag/fourcc.
+@item -qphist
+Show QP histogram.
+@item -vbsf @var{bitstream_filter}
+Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump".
+@example
+ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
+@end example
+@end table
+
+@section Audio Options
+
+@table @option
+@item -aframes @var{number}
+Set the number of audio frames to record.
+@item -ar @var{freq}
+Set the audio sampling frequency (default = 44100 Hz).
+@item -ab @var{bitrate}
+Set the audio bitrate in bit/s (default = 64k).
+@item -aq @var{q}
+Set the audio quality (codec-specific, VBR).
+@item -ac @var{channels}
+Set the number of audio channels (default = 1).
+@item -an
+Disable audio recording.
+@item -acodec @var{codec}
+Force audio codec to @var{codec}. Use the @code{copy} special value to
+specify that the raw codec data must be copied as is.
+@item -newaudio
+Add a new audio track to the output file. If you want to specify parameters,
+do so before @code{-newaudio} (@code{-acodec}, @code{-ab}, etc..).
+
+Mapping will be done automatically, if the number of output streams is equal to
+the number of input streams, else it will pick the first one that matches. You
+can override the mapping using @code{-map} as usual.
+
+Example:
+@example
+ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
+@end example
+@item -alang @var{code}
+Set the ISO 639 language code (3 letters) of the current audio stream.
+@end table
+
+@section Advanced Audio options:
+
+@table @option
+@item -atag @var{fourcc/tag}
+Force audio tag/fourcc.
+@item -absf @var{bitstream_filter}
+Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
+@end table
+
+@section Subtitle options:
+
+@table @option
+@item -scodec @var{codec}
+Force subtitle codec ('copy' to copy stream).
+@item -newsubtitle
+Add a new subtitle stream to the current output stream.
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
+@item -sn
+Disable subtitle recording.
+@item -sbsf @var{bitstream_filter}
+Bitstream filters available are "mov2textsub", "text2movsub".
+@example
+ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
+@end example
+@end table
+
+@section Audio/Video grab options
+
+@table @option
+@item -vc @var{channel}
+Set video grab channel (DV1394 only).
+@item -tvstd @var{standard}
+Set television standard (NTSC, PAL (SECAM)).
+@item -isync
+Synchronize read on input.
+@end table
+
+@section Advanced options
+
+@table @option
+@item -map @var{input_stream_id}[:@var{sync_stream_id}]
+Set stream mapping from input streams to output streams.
+Just enumerate the input streams in the order you want them in the output.
+@var{sync_stream_id} if specified sets the input stream to sync
+against.
+@item -map_meta_data @var{outfile}:@var{infile}
+Set meta data information of @var{outfile} from @var{infile}.
+@item -debug
+Print specific debug info.
+@item -benchmark
+Show benchmarking information at the end of an encode.
+Shows CPU time used and maximum memory consumption.
+Maximum memory consumption is not supported on all systems,
+it will usually display as 0 if not supported.
+@item -dump
+Dump each input packet.
+@item -hex
+When dumping packets, also dump the payload.
+@item -bitexact
+Only use bit exact algorithms (for codec testing).
+@item -ps @var{size}
+Set RTP payload size in bytes.
+@item -re
+Read input at native frame rate. Mainly used to simulate a grab device.
+@item -loop_input
+Loop over the input stream. Currently it works only for image
+streams. This option is used for automatic FFserver testing.
+@item -loop_output @var{number_of_times}
+Repeatedly loop output for formats that support looping such as animated GIF
+(0 will loop the output infinitely).
+@item -threads @var{count}
+Thread count.
+@item -vsync @var{parameter}
+Video sync method.
+0   Each frame is passed with its timestamp from the demuxer to the muxer
+1   Frames will be duplicated and dropped to achieve exactly the requested
+    constant framerate.
+2   Frames are passed through with their timestamp or dropped so as to prevent
+    2 frames from having the same timestamp
+-1  Chooses between 1 and 2 depending on muxer capabilities. This is the default method.
+
+With -map you can select from
+which stream the timestamps should be taken. You can leave either video or
+audio unchanged and sync the remaining stream(s) to the unchanged one.
+@item -async @var{samples_per_second}
+Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
+the parameter is the maximum samples per second by which the audio is changed.
+-async 1 is a special case where only the start of the audio stream is corrected
+without any later correction.
+@item -copyts
+Copy timestamps from input to output.
+@item -shortest
+Finish encoding when the shortest input stream ends.
+@item -dts_delta_threshold
+Timestamp discontinuity delta threshold.
+@item -muxdelay @var{seconds}
+Set the maximum demux-decode delay.
+@item -muxpreload @var{seconds}
+Set the initial demux-decode delay.
+@end table
+
+@section Preset files
+
+A preset file contains a sequence of @var{option}=@var{value} pairs,
+one for each line, specifying a sequence of options which would be
+awkward to specify on the command line. Lines starting with the hash
+('#') character are ignored and are used to provide comments. Check
+the @file{ffpresets} directory in the FFmpeg source tree for examples.
+
+Preset files are specified with the @code{vpre}, @code{apre},
+@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
+filename of the preset instead of a preset name as input and can be
+used for any kind of codec. For the @code{vpre}, @code{apre}, and
+@code{spre} options, the options specified in a preset file are
+applied to the currently selected codec of the same type as the preset
+option.
+
+The argument passed to the @code{vpre}, @code{apre}, and @code{spre}
+preset options identifies the preset file to use according to the
+following rules:
+
+First ffmpeg searches for a file named @var{arg}.ffpreset in the
+directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
+the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg})
+in that order. For example, if the argument is @code{libx264-max}, it will
+search for the file @file{libx264-max.ffpreset}.
+
+If no such file is found, then ffmpeg will search for a file named
+@var{codec_name}-@var{arg}.ffpreset in the above-mentioned
+directories, where @var{codec_name} is the name of the codec to which
+the preset file options will be applied. For example, if you select
+the video codec with @code{-vcodec libx264} and use @code{-vpre max},
+then it will search for the file @file{libx264-max.ffpreset}.
+
+@anchor{FFmpeg formula evaluator}
+@section FFmpeg formula evaluator
+
+When evaluating a rate control string, FFmpeg uses an internal formula
+evaluator.
+
+The following binary operators are available: @code{+}, @code{-},
+@code{*}, @code{/}, @code{^}.
+
+The following unary operators are available: @code{+}, @code{-},
+@code{(...)}.
+
+The following statements are available: @code{ld}, @code{st},
+@code{while}.
+
+The following functions are available:
+@table @var
+@item sinh(x)
+@item cosh(x)
+@item tanh(x)
+@item sin(x)
+@item cos(x)
+@item tan(x)
+@item atan(x)
+@item asin(x)
+@item acos(x)
+@item exp(x)
+@item log(x)
+@item abs(x)
+@item squish(x)
+@item gauss(x)
+@item mod(x, y)
+@item max(x, y)
+@item min(x, y)
+@item eq(x, y)
+@item gte(x, y)
+@item gt(x, y)
+@item lte(x, y)
+@item lt(x, y)
+@item bits2qp(bits)
+@item qp2bits(qp)
+@end table
+
+The following constants are available:
+@table @var
+@item PI
+@item E
+@item iTex
+@item pTex
+@item tex
+@item mv
+@item fCode
+@item iCount
+@item mcVar
+@item var
+@item isI
+@item isP
+@item isB
+@item avgQP
+@item qComp
+@item avgIITex
+@item avgPITex
+@item avgPPTex
+@item avgBPTex
+@item avgTex
+@end table
+
+@c man end
+
+@ignore
+
+@setfilename ffmpeg
+@settitle FFmpeg video converter
+
+@c man begin SEEALSO
+ffserver(1), ffplay(1) and the HTML documentation of @file{ffmpeg}.
+@c man end
+
+@c man begin AUTHOR
+Fabrice Bellard
+@c man end
+
+@end ignore
+
+@section Protocols
+
+The file name can be @file{-} to read from standard input or to write
+to standard output.
+
+FFmpeg also handles many protocols specified with an URL syntax.
+
+Use 'ffmpeg -protocols' to see a list of the supported protocols.
+
+The protocol @code{http:} is currently used only to communicate with
+FFserver (see the FFserver documentation). When FFmpeg will be a
+video player it will also be used for streaming :-)
+
+@chapter Tips
+
+@itemize
+@item For streaming at very low bitrate application, use a low frame rate
+and a small GOP size. This is especially true for RealVideo where
+the Linux player does not seem to be very fast, so it can miss
+frames. An example is:
+
+@example
+ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
+@end example
+
+@item  The parameter 'q' which is displayed while encoding is the current
+quantizer. The value 1 indicates that a very good quality could
+be achieved. The value 31 indicates the worst quality. If q=31 appears
+too often, it means that the encoder cannot compress enough to meet
+your bitrate. You must either increase the bitrate, decrease the
+frame rate or decrease the frame size.
+
+@item If your computer is not fast enough, you can speed up the
+compression at the expense of the compression ratio. You can use
+'-me zero' to speed up motion estimation, and '-intra' to disable
+motion estimation completely (you have only I-frames, which means it
+is about as good as JPEG compression).
+
+@item To have very low audio bitrates, reduce the sampling frequency
+(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
+
+@item To have a constant quality (but a variable bitrate), use the option
+'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
+quality).
+
+@item When converting video files, you can use the '-sameq' option which
+uses the same quality factor in the encoder as in the decoder.
+It allows almost lossless encoding.
+
+@end itemize
+
+@bye

doc/ffmpeg-filters.texi

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

doc/ffmpeg-formats.texi

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

doc/ffmpeg-protocols.texi

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

doc/ffmpeg-resampler.texi

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

doc/ffmpeg-scaler.texi

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

doc/ffmpeg-utils.texi

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

doc/ffmpeg.txt

@@ -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                      :
-                                                                                                                                       :

doc/ffmpeg_powerpc_performance_evaluation_howto.txt

@@ -0,0 +1,172 @@
+FFmpeg & evaluating performance on the PowerPC Architecture HOWTO
+
+(c) 2003-2004 Romain Dolbeau <romain@dolbeau.org>
+
+
+
+I - Introduction
+
+The PowerPC architecture and its SIMD extension AltiVec offer some
+interesting tools to evaluate performance and improve the code.
+This document tries to explain how to use those tools with FFmpeg.
+
+The architecture itself offers two ways to evaluate the performance of
+a given piece of code:
+
+1) The Time Base Registers (TBL)
+2) The Performance Monitor Counter Registers (PMC)
+
+The first ones are always available, always active, but they're not very
+accurate: the registers increment by one every four *bus* cycles. On
+my 667 Mhz tiBook (ppc7450), this means once every twenty *processor*
+cycles. So we won't use that.
+
+The PMC are much more useful: not only can they report cycle-accurate
+timing, but they can also be used to monitor many other parameters,
+such as the number of AltiVec stalls for every kind of instruction,
+or instruction cache misses. The downside is that not all processors
+support the PMC (all G3, all G4 and the 970 do support them), and
+they're inactive by default - you need to activate them with a
+dedicated tool. Also, the number of available PMC depends on the
+procesor: the various 604 have 2, the various 75x (aka. G3) have 4,
+and the various 74xx (aka G4) have 6.
+
+*WARNING*: The PowerPC 970 is not very well documented, and its PMC
+registers are 64 bits wide. To properly notify the code, you *must*
+tune for the 970 (using --tune=970), or the code will assume 32 bit
+registers.
+
+
+II - Enabling FFmpeg PowerPC performance support
+
+This needs to be done by hand. First, you need to configure FFmpeg as
+usual, but add the "--powerpc-perf-enable" option. For instance:
+
+#####
+./configure --prefix=/usr/local/ffmpeg-svn --cc=gcc-3.3 --tune=7450 --powerpc-perf-enable
+#####
+
+This will configure FFmpeg to install inside /usr/local/ffmpeg-svn,
+compiling with gcc-3.3 (you should try to use this one or a newer
+gcc), and tuning for the PowerPC 7450 (i.e. the newer G4; as a rule of
+thumb, those at 550Mhz and more). It will also enable the PMC.
+
+You may also edit the file "config.h" to enable the following line:
+
+#####
+// #define ALTIVEC_USE_REFERENCE_C_CODE 1
+#####
+
+If you enable this line, then the code will not make use of AltiVec,
+but will use the reference C code instead. This is useful to compare
+performance between two versions of the code.
+
+Also, the number of enabled PMC is defined in "libavcodec/ppc/dsputil_ppc.h":
+
+#####
+#define POWERPC_NUM_PMC_ENABLED 4
+#####
+
+If you have a G4 CPU, you can enable all 6 PMC. DO NOT enable more
+PMC than available on your CPU!
+
+Then, simply compile FFmpeg as usual (make && make install).
+
+
+
+III - Using FFmpeg PowerPC performance support
+
+This FFmeg can be used exactly as usual. But before exiting, FFmpeg
+will dump a per-function report that looks like this:
+
+#####
+PowerPC performance report
+ Values are from the PMC registers, and represent whatever the
+ registers are set to record.
+ Function "gmc1_altivec" (pmc1):
+        min: 231
+        max: 1339867
+        avg: 558.25 (255302)
+ Function "gmc1_altivec" (pmc2):
+        min: 93
+        max: 2164
+        avg: 267.31 (255302)
+ Function "gmc1_altivec" (pmc3):
+        min: 72
+        max: 1987
+        avg: 276.20 (255302)
+(...)
+#####
+
+In this example, PMC1 was set to record CPU cycles, PMC2 was set to
+record AltiVec Permute Stall Cycles, and PMC3 was set to record AltiVec
+Issue Stalls.
+
+The function "gmc1_altivec" was monitored 255302 times, and the
+minimum execution time was 231 processor cycles. The max and average
+aren't much use, as it's very likely the OS interrupted execution for
+reasons of its own :-(
+
+With the exact same settings and source file, but using the reference C
+code we get:
+
+#####
+PowerPC performance report
+ Values are from the PMC registers, and represent whatever the
+ registers are set to record.
+ Function "gmc1_altivec" (pmc1):
+        min: 592
+        max: 2532235
+        avg: 962.88 (255302)
+ Function "gmc1_altivec" (pmc2):
+        min: 0
+        max: 33
+        avg: 0.00 (255302)
+ Function "gmc1_altivec" (pmc3):
+        min: 0
+        max: 350
+        avg: 0.03 (255302)
+(...)
+#####
+
+592 cycles, so the fastest AltiVec execution is about 2.5x faster than
+the fastest C execution in this example. It's not perfect but it's not
+bad (well I wrote this function so I can't say otherwise :-).
+
+Once you have that kind of report, you can try to improve things by
+finding what goes wrong and fixing it; in the example above, one
+should try to diminish the number of AltiVec stalls, as this *may*
+improve performance.
+
+
+
+IV) Enabling the PMC in Mac OS X
+
+This is easy. Use "Monster" and "monster". Those tools come from
+Apple's CHUD package, and can be found hidden in the developer web
+site & FTP site. "MONster" is the graphical application, use it to
+generate a config file specifying what each register should
+monitor. Then use the command-line application "monster" to use that
+config file, and enjoy the results.
+
+Note that "MONster" can be used for many other things, but it's
+documented by Apple, it's not my subject.
+
+If you are using CHUD 4.4.2 or later, you'll notice that MONster is
+no longer available. It's been superseeded by Shark, where
+configuration of PMCs is available as a plugin.
+
+
+
+V) Enabling the PMC on Linux
+
+On linux you may use oprofile from http://oprofile.sf.net, depending on the
+version and the cpu you may need to apply a patch[1] to access a set of the
+possibile counters from the userspace application. You can always define them
+using the kernel interface /dev/oprofile/* .
+
+[1] http://dev.gentoo.org/~lu_zero/development/oprofile-g4-20060423.patch
+
+--
+Romain Dolbeau <romain@dolbeau.org>
+Luca Barbato <lu_zero@gentoo.org>

doc/ffplay-doc.texi

@@ -1,30 +1,33 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle ffplay Documentation
+@settitle FFplay Documentation
 @titlepage
-@center @titlefont{ffplay Documentation}
+@sp 7
+@center @titlefont{FFplay Documentation}
+@sp 3
 @end titlepage
 
-@top
 
-@contents
+@chapter Introduction
 
-@chapter Synopsis
-
-ffplay [@var{options}] [@file{input_file}]
-
-@chapter Description
 @c man begin DESCRIPTION
-
 FFplay is a very simple and portable media player using the FFmpeg
 libraries and the SDL library. It is mostly used as a testbed for the
 various FFmpeg APIs.
 @c man end
 
-@chapter Options
+@chapter Invocation
+
+@section Syntax
+@example
+@c man begin SYNOPSIS
+ffplay [options] @file{input_file}
+@c man end
+@end example
+
 @c man begin OPTIONS
 
-@include avtools-common-opts.texi
+@include fftools-common-opts.texi
 
 @section Main options
 
@@ -34,9 +37,8 @@ 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
@@ -55,44 +57,21 @@ Force format.
 Set window title (default is the input filename).
 @item -loop @var{number}
 Loops movie playback <number> times. 0 means forever.
-@item -showmode @var{mode}
-Set the show mode to use.
-Available values for @var{mode} are:
-@table @samp
-@item 0, video
-show video
-@item 1, waves
-show audio waves
-@item 2, rdft
-show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
-@end table
-
-Default value is "video", if video is not present or cannot be played
-"rdft" is automatically selected.
-
-You can interactively cycle through the available show modes by
-pressing the key @key{w}.
-
-@item -vf @var{filter_graph}
-@var{filter_graph} is a description of the filter graph to apply to
-the input video.
-Use the option "-filters" to show all the available filters (including
-also sources and sinks).
-
-@item -i @var{input_file}
-Read @var{input_file}.
 @end table
 
 @section Advanced options
 @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.
+@item -debug
+Print specific debug info.
 @item -bug
 Work around bugs.
+@item -vismv
+Visualize motion vectors.
 @item -fast
 Non-spec-compliant optimizations.
 @item -genpts
@@ -124,26 +103,6 @@ Select the desired subtitle stream number, counting from 0. The number
 refers to the list of all the input subtitle streams. If it is greater
 than the number of subtitle streams minus one, then the last one is
 selected, if it is negative the subtitle rendering is disabled.
-@item -autoexit
-Exit when video is done playing.
-@item -exitonkeydown
-Exit if any key is pressed.
-@item -exitonmousedown
-Exit if any mouse button is pressed.
-
-@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 +135,6 @@ 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 +142,19 @@ 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
-
 @ignore
 
 @setfilename ffplay
 @settitle FFplay media player
 
+@c man begin SEEALSO
+ffmpeg(1), ffserver(1) and the HTML documentation of @file{ffmpeg}.
+@c man end
+
+@c man begin AUTHOR
+Fabrice Bellard
+@c man end
+
 @end ignore
 
 @bye

doc/ffprobe-doc.texi

@@ -0,0 +1,121 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFprobe Documentation
+@titlepage
+@sp 7
+@center @titlefont{FFprobe Documentation}
+@sp 3
+@end titlepage
+
+@chapter Introduction
+
+@c man begin DESCRIPTION
+
+FFprobe gathers information from multimedia streams and prints it in
+human- and machine-readable fashion.
+
+For example it can be used to check the format of the container used
+by a multimedia stream and the format and type of each media stream
+contained in it.
+
+If a filename is specified in input, ffprobe will try to open and
+probe the file content. If the file cannot be opened or recognized as
+a multimedia file, a positive exit code is returned.
+
+FFprobe may be employed both as a standalone application or in
+combination with a textual filter, which may perform more
+sophisticated processing, e.g. statistical processing or plotting.
+
+Options are used to list some of the formats supported by ffprobe or
+for specifying which information to display, and for setting how
+ffprobe will show it.
+
+FFprobe output is designed to be easily parsable by a textual filter,
+and consists of one or more sections of the form:
+@example
+[SECTION]
+key1=val1
+...
+keyN=valN
+[/SECTION]
+@end example
+
+Metadata tags stored in the container or in the streams are recognized
+and printed in the corresponding ``FORMAT'' or ``STREAM'' section, and
+are prefixed by the string ``TAG:''.
+
+@c man end
+
+@chapter Invocation
+
+@section Syntax
+
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffprobe [options] [@file{input_file}]
+@c man end
+@end example
+
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+
+@item -convert_tags
+Convert the tag names in the format container to the generic FFmpeg tag names.
+
+@item -f @var{format}
+Force format to use.
+
+@item -unit
+Show the unit of the displayed values.
+
+@item -prefix
+Show a SI prefixes of the displayed values.
+Unless ``-byte_binary_prefix'' option is used all the prefix
+are decimal.
+
+@item -byte_binary_prefix
+Force the use of binary prefixes for byte values.
+
+@item -sexagesimal
+Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
+
+@item -pretty
+Prettify the format of the displayed values, it corresponds to the
+options ``-unit -prefix -byte_binary_prefix -sexagesimal''.
+
+@item -show_format
+Show information about the container format of the input multimedia
+stream.
+
+All the container format information is printed within a section with
+name ``FORMAT''.
+
+@item -show_streams
+Show information about each media stream contained in the input
+multimedia stream.
+
+Each media stream information is printed within a dedicated section
+with name ``STREAM''.
+
+@end table
+@c man end
+
+@ignore
+
+@setfilename ffprobe
+@settitle FFprobe media prober
+
+@c man begin SEEALSO
+ffmpeg(1), ffplay(1), ffserver(1)
+@c man end
+
+@end ignore
+
+@bye

doc/ffprobe.texi

@@ -1,521 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle ffprobe Documentation
-@titlepage
-@center @titlefont{ffprobe Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Synopsis
-
-ffprobe [@var{options}] [@file{input_file}]
-
-@chapter Description
-@c man begin DESCRIPTION
-
-ffprobe gathers information from multimedia streams and prints it in
-human- and machine-readable fashion.
-
-For example it can be used to check the format of the container used
-by a multimedia stream and the format and type of each media stream
-contained in it.
-
-If a filename is specified in input, ffprobe will try to open and
-probe the file content. If the file cannot be opened or recognized as
-a multimedia file, a positive exit code is returned.
-
-ffprobe may be employed both as a standalone application or in
-combination with a textual filter, which may perform more
-sophisticated processing, e.g. statistical processing or plotting.
-
-Options are used to list some of the formats supported by ffprobe or
-for specifying which information to display, and for setting how
-ffprobe will show it.
-
-ffprobe output is designed to be easily parsable by a textual filter,
-and consists of one or more sections of a form defined by the selected
-writer, which is specified by the @option{print_format} option.
-
-Sections may contain other nested sections, and are identified by a
-name (which may be shared by other sections), and an unique
-name. See the output of @option{sections}.
-
-Metadata tags stored in the container or in the streams are recognized
-and printed in the corresponding "FORMAT" or "STREAM" section.
-
-@c man end
-
-@chapter Options
-@c man begin OPTIONS
-
-@include avtools-common-opts.texi
-
-@section Main options
-
-@table @option
-
-@item -f @var{format}
-Force format to use.
-
-@item -unit
-Show the unit of the displayed values.
-
-@item -prefix
-Use SI prefixes for the displayed values.
-Unless the "-byte_binary_prefix" option is used all the prefixes
-are decimal.
-
-@item -byte_binary_prefix
-Force the use of binary prefixes for byte values.
-
-@item -sexagesimal
-Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
-
-@item -pretty
-Prettify the format of the displayed values, it corresponds to the
-options "-unit -prefix -byte_binary_prefix -sexagesimal".
-
-@item -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.
-
-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.
-
-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.
-
-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
-
-@ignore
-
-@setfilename ffprobe
-@settitle ffprobe media prober
-
-@end ignore
-
-@bye

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>

doc/ffserver-doc.texi

@@ -1,35 +1,34 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle ffserver Documentation
+@settitle FFserver Documentation
 @titlepage
-@center @titlefont{ffserver Documentation}
+@sp 7
+@center @titlefont{FFserver Documentation}
+@sp 3
 @end titlepage
 
-@top
 
-@contents
+@chapter Introduction
 
-@chapter Synopsis
-
-ffserver [@var{options}]
-
-@chapter Description
 @c man begin DESCRIPTION
-
-ffserver is a streaming server for both audio and video. It supports
-
+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?
 
-ffserver receives prerecorded files or FFM streams from some ffmpeg
+FFserver receives prerecorded files or FFM streams from some ffmpeg
 instance as input, then streams them over RTP/RTSP/HTTP.
 
 An ffserver instance will listen on some port as specified in the
@@ -47,7 +46,7 @@ file.
 
 @section Status stream
 
-ffserver supports an HTTP interface which exposes the current status
+FFserver supports an HTTP interface which exposes the current status
 of the server.
 
 Simply point your browser to the address of the special status stream
@@ -81,7 +80,7 @@ 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 +99,8 @@ 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 +135,7 @@ 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 +226,18 @@ 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
+@chapter Invocation
+@section Syntax
+@example
+@c man begin SYNOPSIS
+ffserver [options]
+@c man end
+@end example
 
-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
+@section Options
 @c man begin OPTIONS
 
-@include avtools-common-opts.texi
+@include fftools-common-opts.texi
 
 @section Main options
 
@@ -252,43 +246,28 @@ differing versions of tools. FFM2 is the default.
 Use @file{configfile} instead of @file{/etc/ffserver.conf}.
 @item -n
 Enable no-launch mode. This option disables all the Launch directives
-within the various <Stream> sections. Since ffserver will not launch
-any ffmpeg instances, you will have to launch them manually.
+within the various <Stream> sections. FFserver will not launch any
+ffmpeg instance, so you will have to launch them manually.
 @item -d
 Enable debug mode. This option increases log verbosity, directs log
-messages to stdout.
+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
+@settitle FFserver video server
+
+@c man begin SEEALSO
+ffmpeg(1), ffplay(1), the @file{ffmpeg/doc/ffserver.conf} example and
+the HTML documentation of @file{ffmpeg}.
+@c man end
+
+@c man begin AUTHOR
+Fabrice Bellard
+@c man end
 
 @end ignore
 

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
@@ -302,27 +306,6 @@ StartSendOnKey
 #</Stream>
 
 
-# Transcode an incoming live feed to another live feed,
-# using libx264 and video presets
-
-#<Stream live.h264>
-#Format rtp
-#Feed feed1.ffm
-#VideoCodec libx264
-#VideoFrameRate 24
-#VideoBitRate 100
-#VideoSize 480x272
-#AVPresetVideo default
-#AVPresetVideo baseline
-#AVOptionVideo flags +global_header
-#
-#AudioCodec libfaac
-#AudioBitRate 32
-#AudioChannels 2
-#AudioSampleRate 22050
-#AVOptionAudio flags +global_header
-#</Stream>
-
 ##################################################################
 # SDP/multicast examples
 #
@@ -369,3 +352,5 @@ ACL allow 192.168.0.0 192.168.255.255
 <Redirect index.html>
 URL http://www.ffmpeg.org/
 </Redirect>
+
+

doc/fftools-common-opts.texi

@@ -0,0 +1,72 @@
+@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
+
+@end table

doc/filter_design.txt

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

doc/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.

doc/git-howto.txt

@@ -1,273 +0,0 @@
-
-About Git write access:
-~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-
-Before everything else, you should know how to use GIT properly.
-Luckily Git comes with excellent documentation.
-
-  git --help
-  man git
-
-shows you the available subcommands,
-
-  git <command> --help
-  man git-<command>
-
-shows information about the subcommand <command>.
-
-The most comprehensive manual is the website Git Reference
-
-http://gitref.org/
-
-For more information about the Git project, visit
-
-http://git-scm.com/
-
-Consult these resources whenever you have problems, they are quite exhaustive.
-
-You do not need a special username or password.
-All you need is to provide a ssh public key to the Git server admin.
-
-What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines. Read it at least once, if you are granted commit privileges to the
-FFmpeg project you are expected to be familiar with these rules.
-
-
-
-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>
-
-  This will put the FFmpeg sources into the directory <target>.
-
-    git clone git@source.ffmpeg.org:ffmpeg <target>
-
-  This will put the FFmpeg sources into the directory <target> and let
-  you push back your changes to the remote repository.
-
-
-2. Updating the source tree to the latest revision:
-
-    git pull (--ff-only)
-
-  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.
-  Caveat: Since merge commits are forbidden at least for the initial
-          months of git --ff-only or --rebase (see below) are recommended.
-          --ff-only will fail and not create merge commits if your branch
-          has diverged (has a different history) from the tracked branch.
-
-2.a Rebasing your local branches:
-
-    git pull --rebase
-
-  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.
-
-
-3. Adding/removing files/directories:
-
-    git add [-A] <filename/dirname>
-    git rm [-r] <filename/dirname>
-
-  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.
-
-
-4. Showing modifications:
-
-    git diff <filename(s)>
-
-  will show all local modifications in your working directory as unified diff.
-
-
-5. Inspecting the changelog:
-
-    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
-
-6. Checking source tree status:
-
-    git status
-
-  detects all the changes you made and lists what actions will be taken in case
-  of a commit (additions, modifications, deletions, etc.).
-
-
-7. Committing:
-
-    git diff --check
-
-  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
-
-    git diff -w -b <filename(s)>
-
-  Also check the output of
-
-    git status
-
-  to make sure you don't have untracked files or deletions.
-
-    git add [-i|-p|-A] <filenames/dirnames>
-
-  Make sure you have told git your name and email address, e.g. by running
-    git config --global user.name "My Name"
-    git config --global user.email my@email.invalid
-  (--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.
-
-    git commit
-
-  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
-
-    git config core.editor
-
-  or set by one of the following environment variables:
-  GIT_EDITOR, VISUAL or 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.
-
-
-8. Renaming/moving/copying files or contents of files:
-
-  Git automatically tracks such changes, making those normal commits.
-
-    mv/cp path/file otherpath/otherfile
-
-    git add [-A] .
-
-    git commit
-
-  Do not move, rename or copy files of which you are not the maintainer without
-  discussing it on the mailing list first!
-
-9. Reverting broken commits
-
-    git revert <commit>
-
-  git revert will generate a revert commit. This will not make the faulty
-  commit disappear from the history.
-
-    git reset <commit>
-
-  git reset will uncommit the changes till <commit> rewriting the current
-  branch history.
-
-    git commit --amend
-
-  allows to amend the last commit details quickly.
-
-    git rebase -i origin/master
-
-  will replay local commits over the main repository allowing to edit,
-  merge or remove some of them in the process.
-
-  Note that the reset, commit --amend and rebase rewrite history, so you
-  should use them ONLY on your local or topic branches.
-
-  The main repository will reject those changes.
-
-10. Preparing a patchset.
-
-    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.
-
-11. Sending patches for review
-
-    git send-email <commit list|directory>
-
-  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
-
-    git push
-
-  Will push the changes to the default remote (origin).
-  Git will prevent you from pushing changes if the local and remote trees are
-  out of sync. Refer to 2 and 2.a to sync the local tree.
-
-    git remote add <name> <url>
-
-  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.
-
-    git push <remote> <refspec>
-
-  Will push the changes to the remote repository. Omitting refspec makes git
-  push update all the remote branches matching the local ones.
-
-13. Finding a specific svn revision
-
-  Since version 1.7.1 git supports ':/foo' syntax for specifying commits
-  based on a regular expression. see man gitrevisions
-
-    git show :/'as revision 23456'
-
-  will show the svn changeset r23456. With older git versions searching in
-  the git log output is the easiest option (especially if a pager with
-  search capabilities is used).
-  This commit can be checked out with
-
-    git checkout -b svn_23456 :/'as revision 23456'
-
-  or for git < 1.7.1 with
-
-    git checkout -b svn_23456 $SHA1
-
-  where $SHA1 is the commit SHA1 from the 'git log' output.
-
-
-Contact the project admins <root at ffmpeg dot org> if you have technical
-problems with the GIT server.

doc/indevs.texi

@@ -1,717 +0,0 @@
-@chapter Input Devices
-@c man begin INPUT DEVICES
-
-Input devices are configured elements in FFmpeg which allow to access
-the data coming from a multimedia device attached to your system.
-
-When you configure your FFmpeg build, all the supported input devices
-are enabled by default. You can list all available ones using the
-configure option "--list-indevs".
-
-You can disable all the input devices using the configure option
-"--disable-indevs", and selectively enable an input device using the
-option "--enable-indev=@var{INDEV}", or you can disable a particular
-input device using the option "--disable-indev=@var{INDEV}".
-
-The option "-formats" of the ff* tools will display the list of
-supported input devices (amongst the demuxers).
-
-A description of the currently available input devices follows.
-
-@section alsa
-
-ALSA (Advanced Linux Sound Architecture) input device.
-
-To enable this input device during configuration you need libasound
-installed on your system.
-
-This device allows capturing from an ALSA device. The name of the
-device to capture has to be an ALSA card identifier.
-
-An ALSA identifier has the syntax:
-@example
-hw:@var{CARD}[,@var{DEV}[,@var{SUBDEV}]]
-@end example
-
-where the @var{DEV} and @var{SUBDEV} components are optional.
-
-The three arguments (in order: @var{CARD},@var{DEV},@var{SUBDEV})
-specify card number or identifier, device number and subdevice number
-(-1 means any).
-
-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
-card id 0, you may run the command:
-@example
-ffmpeg -f alsa -i hw:0 alsaout.wav
-@end example
-
-For more information see:
-@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
-
-@section bktr
-
-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.
-
-@section fbdev
-
-Linux framebuffer input device.
-
-The Linux framebuffer is a graphic hardware-independent abstraction
-layer to show graphics on a computer monitor, typically on the
-console. It is accessed through a file device node, usually
-@file{/dev/fb0}.
-
-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}:
-@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
-@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.
-
-To enable this input device during configuration you need libjack
-installed on your system.
-
-A JACK input device creates one or more JACK writable clients, one for
-each audio channel, with name @var{client_name}:input_@var{N}, where
-@var{client_name} is the name provided by the application, and @var{N}
-is a number which identifies the channel.
-Each writable client will send the acquired data to the FFmpeg input
-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 list the JACK clients and their properties you can invoke the command
-@command{jack_lsp}.
-
-Follows an example which shows how to capture a JACK readable client
-with @command{ffmpeg}.
-@example
-# Create a JACK writable client with name "ffmpeg".
-$ ffmpeg -f jack -i ffmpeg -y out.wav
-
-# Start the sample jack_metro readable client.
-$ jack_metro -b 120 -d 0.2 -f 4000
-
-# List the current JACK clients.
-$ jack_lsp -c
-system:capture_1
-system:capture_2
-system:playback_1
-system:playback_2
-ffmpeg:input_1
-metro:120_bpm
-
-# Connect metro to the ffmpeg writable client.
-$ jack_connect metro:120_bpm ffmpeg:input_1
-@end example
-
-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.
-
-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
-command:
-@example
-ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
-@end example
-
-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.
-
-To enable this input device during configuration you need libsndio
-installed on your system.
-
-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
-command:
-@example
-ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
-@end example
-
-@section video4linux2
-
-Video4Linux2 input video device.
-
-The name of the device to grab is a file device node, usually Linux
-systems tend to automatically create such nodes when the device
-(e.g. an USB webcam) is plugged into the system, and has a name of the
-kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
-the device.
-
-Video4Linux2 devices usually 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.
-
-Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
-
-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.
-
-Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
-option), it will always be used.
-@example
-# Grab and show the input of a video4linux2 device.
-ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
-
-# Grab and record the input of a video4linux2 device, leave the
-framerate and size as previously set.
-ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
-@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.
-
-The filename passed as input is the capture driver number, ranging from
-0 to 9. You may use "list" as filename to print a list of drivers. Any
-other filename will be interpreted as device number 0.
-
-@section x11grab
-
-X11 video input device.
-
-This device allows to capture a region of an X11 display.
-
-The filename passed as input has the syntax:
-@example
-[@var{hostname}]:@var{display_number}.@var{screen_number}[+@var{x_offset},@var{y_offset}]
-@end example
-
-@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
-@env{DISPLAY} contains the default display name.
-
-@var{x_offset} and @var{y_offset} specify the offsets of the grabbed
-area with respect to the top-left border of the X11 screen. They
-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
-properties of your X11 display (e.g. grep for "name" or "dimensions").
-
-For example to grab from @file{:0.0} using @command{ffmpeg}:
-@example
-ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
-@end example
-
-Grab at position @code{10,20}:
-@example
-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

doc/issue_tracker.txt

@@ -5,42 +5,49 @@ 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.
-
+FFmpeg uses Roundup for tracking issues, new issues and changes to
+existing issues can be done through a web interface and through email.
 It is possible to subscribe to individual issues by adding yourself to the
-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 subscription URL for the ffmpeg-issues list is:
+http://live.polito/mailman/listinfo/ffmpeg-issues
 The URL of the webinterface of the tracker is:
-http(s)://trac.ffmpeg.org
+http(s)://roundup.ffmpeg/roundup/ffmpeg/
+Note the URLs in this document are obfuscated, you must append the top level
+domain for non-profit organizations to the tracker, and of Italy to the
+mailing list.
+
+Email Interface:
+----------------
+There is a mailing list to which all new issues and changes to existing issues
+are sent. You can subscribe through
+http://live.polito/mailman/listinfo/ffmpeg-issues
+Replies to messages there will have their text added to the specific issues.
+Attachments will be added as if they had been uploaded via the web interface.
+You can change the status, substatus, topic, ... by changing the subject in
+your reply like:
+Re: [issue94] register_avcodec and allcodecs.h [type=patch;status=open;substatus=approved]
+Roundup will then change things as you requested and remove the [...] from
+the subject before forwarding the mail to the mailing list.
+
+
+NOTE: issue = (bug report || patch || feature request)
 
 Type:
 -----
-bug / 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 +68,6 @@ 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 +97,6 @@ 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 +124,24 @@ 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 +175,10 @@ 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 +190,12 @@ 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 +215,6 @@ 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 +222,7 @@ 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

doc/libavcodec.texi

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

doc/libavdevice.texi

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

doc/libavfilter.texi

@@ -2,43 +2,306 @@
 
 @settitle Libavfilter Documentation
 @titlepage
+@sp 7
 @center @titlefont{Libavfilter Documentation}
+@sp 3
 @end titlepage
 
-@top
 
-@contents
+@chapter Introduction
 
-@chapter Description
-@c man begin DESCRIPTION
+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.
 
-The libavfilter library provides a generic audio/video filtering
-framework containing several filters, sources and sinks.
+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:
 
-@c man end DESCRIPTION
+@example
+   svn checkout svn://svn.ffmpeg.org/soc/libavfilter
+@end example
 
-@chapter See Also
+And then read the README file in the top directory to learn how to
+integrate it into ffmpeg and ffplay.
 
-@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
+But note that there may still be serious bugs in the code and its API
+and ABI should not be considered stable yet!
 
-@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
+@chapter Tutorial
 
-@include authors.texi
+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:
 
-@ignore
+@example
+input --> split --> fifo -----------------------> overlay --> output
+            |                                        ^
+            |                                        |
+            +------> fifo --> crop --> vflip --------+
+@end example
 
-@setfilename libavfilter
-@settitle multimedia filtering library
+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:
 
-@end ignore
+@example
+./ffmpeg -i in.avi -s 240x320 -vfilters "[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{-vfilters} 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.
+
+@chapter Available video filters
+
+When you configure your FFmpeg build, you can disable any of the
+existing video filters.
+The configure output will show the video filters included in your
+build.
+
+Below is a description of the currently available video filters.
+
+@section crop
+
+Crop the input video to @var{x}:@var{y}:@var{width}:@var{height}.
+
+@example
+./ffmpeg -i in.avi -vfilters "crop=0:0:0:240" out.avi
+@end example
+
+@var{x} and @var{y} specify the position of the top-left corner of the
+output (non-cropped) area.
+
+The default value of @var{x} and @var{y} is 0.
+
+The @var{width} and @var{height} parameters specify the width and height
+of the output (non-cropped) area.
+
+A value of 0 is interpreted as the maximum possible size contained in
+the area delimited by the top-left corner at position x:y.
+
+For example the parameters:
+
+@example
+"crop=100:100:0:0"
+@end example
+
+will delimit the rectangle with the top-left corner placed at position
+100:100 and the right-bottom corner corresponding to the right-bottom
+corner of the input image.
+
+The default value of @var{width} and @var{height} is 0.
+
+@section format
+
+Convert the input video to one of the specified pixel formats.
+Libavfilter will try to pick one that is supported for the input to
+the next filter.
+
+The filter accepts a list of pixel format names, separated by ``:'',
+for example ``yuv420p:monow:rgb24''.
+
+The following command:
+
+@example
+./ffmpeg -i in.avi -vfilters "format=yuv420p" out.avi
+@end example
+
+will convert the input video to the format ``yuv420p''.
+
+@section noformat
+
+Force libavfilter not to use any of the specified pixel formats for the
+input to the next filter.
+
+The filter accepts a list of pixel format names, separated by ``:'',
+for example ``yuv420p:monow:rgb24''.
+
+The following command:
+
+@example
+./ffmpeg -i in.avi -vfilters "noformat=yuv420p, vflip" out.avi
+@end example
+
+will make libavfilter use a format different from ``yuv420p'' for the
+input to the vflip filter.
+
+@section null
+
+Pass the source unchanged to the output.
+
+@section scale
+
+Scale the input video to @var{width}:@var{height} and/or convert the image format.
+
+For example the command:
+
+@example
+./ffmpeg -i in.avi -vfilters "scale=200:100" out.avi
+@end example
+
+will scale the input video to a size of 200x100.
+
+If the input image format is different from the format requested by
+the next filter, the scale filter will convert the input to the
+requested format.
+
+If the value for @var{width} or @var{height} is 0, the respective input
+size is used for the output.
+
+If the value for @var{width} or @var{height} is -1, the scale filter will
+use, for the respective output size, a value that maintains the aspect
+ratio of the input image.
+
+The default value of @var{width} and @var{height} is 0.
+
+@section slicify
+
+Pass the images of input video on to next video filter as multiple
+slices.
+
+@example
+./ffmpeg -i in.avi -vfilters "slicify=32" out.avi
+@end example
+
+The filter accepts the slice height as parameter. If the parameter is
+not specified it will use the default value of 16.
+
+Adding this in the beginning of filter chains should make filtering
+faster due to better use of the memory cache.
+
+@section unsharp
+
+Sharpen or blur the input video. It accepts the following parameters:
+
+@multitable @columnfractions .2 .5 .1 .1 .1
+@headitem Name @tab Description @tab Min @tab Max @tab Default
+@item @var{luma_msize_x}
+@tab Luma matrix horizontal size
+@tab 3
+@tab 13
+@tab 5
+@item @var{luma_msize_y}
+@tab Luma matrix vertical size
+@tab 3
+@tab 13
+@tab 5
+@item @var{luma_amount}
+@tab Luma effect strength
+@tab -2.0
+@tab 5.0
+@tab 1.0
+@item @var{chroma_msize_x}
+@tab Chroma matrix horizontal size
+@tab 3
+@tab 13
+@tab 0
+@item @var{chroma_msize_y}
+@tab Chroma matrix vertical size
+@tab 3
+@tab 13
+@tab 0
+@item @var{chroma_amount}
+@tab Chroma effect strength
+@tab -2.0
+@tab 5.0
+@tab 0.0
+@end multitable
+
+Negative values for the amount will blur the input video, while positive
+values will sharpen. All parameters are optional and default to the
+equivalent of the string '5:5:1.0:0:0:0.0'.
+
+@example
+# Strong luma sharpen effect parameters
+unsharp=7:7:2.5
+
+# Strong blur of both luma and chroma parameters
+unsharp=7:7:-2:7:7:-2
+
+# Use the default values with @command{ffmpeg}
+./ffmpeg -i in.avi -vfilters "unsharp" out.mp4
+@end example
+
+@section vflip
+
+Flip the input video vertically.
+
+@example
+./ffmpeg -i in.avi -vfilters "vflip" out.avi
+@end example
+
+@chapter Available video sources
+
+Below is a description of the currently available video sources.
+
+@section nullsrc
+
+Null video source, never return images. It is mainly useful as a
+template and to be employed in analysis / debugging tools.
+
+It accepts as optional parameter a string of the form
+@var{width}:@var{height}, where @var{width} and @var{height} specify the size of
+the configured source.
+
+The default values of @var{width} and @var{height} are respectively 352
+and 288 (corresponding to the CIF size format).
+
+@chapter Available video sinks
+
+Below is a description of the currently available video sinks.
+
+@section nullsink
+
+Null video sink, do absolutely nothing with the input video. It is
+mainly useful as a template and to be employed in analysis / debugging
+tools.
 
 @bye

doc/libavformat.texi

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

doc/libavutil.texi

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

doc/libswresample.texi

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

doc/libswscale.texi

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

doc/metadata.texi

@@ -1,68 +0,0 @@
-@chapter Metadata
-@c man begin METADATA
-
-FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded
-INI-like text file and then load it back using the metadata muxer/demuxer.
-
-The file format is as follows:
-@enumerate
-
-@item
-A file consists of a header and a number of metadata tags divided into sections,
-each on its own line.
-
-@item
-The header is a ';FFMETADATA' string, followed by a version number (now 1).
-
-@item
-Metadata tags are of the form 'key=value'
-
-@item
-Immediately after header follows global metadata
-
-@item
-After global metadata there may be sections with per-stream/per-chapter
-metadata.
-
-@item
-A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
-brackets ('[', ']') and ends with next section or end of file.
-
-@item
-At the beginning of a chapter section there may be an optional timebase to be
-used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
-den are integers. If the timebase is missing then start/end times are assumed to
-be in milliseconds.
-Next a chapter section must contain chapter start and end times in form
-'START=num', 'END=num', where num is a positive integer.
-
-@item
-Empty lines and lines starting with ';' or '#' are ignored.
-
-@item
-Metadata keys or values containing special characters ('=', ';', '#', '\' and a
-newline) must be escaped with a backslash '\'.
-
-@item
-Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
-the tag (in the example above key is 'foo ', value is ' bar').
-@end enumerate
-
-A ffmetadata file might look like this:
-@example
-;FFMETADATA1
-title=bike\\shed
-;this is a comment
-artist=FFmpeg troll team
-
-[CHAPTER]
-TIMEBASE=1/1000
-START=0
-#chapter ends at 0:01:00
-END=60000
-title=chapter \#1
-[STREAM]
-title=multi\
-line
-@end example
-@c man end METADATA

doc/mips.txt

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

doc/multithreading.txt

@@ -1,65 +0,0 @@
-FFmpeg multithreading methods
-==============================================
-
-FFmpeg provides two methods for multithreading codecs.
-
-Slice threading decodes multiple parts of a frame at the same time, using
-AVCodecContext execute() and execute2().
-
-Frame threading decodes multiple frames at the same time.
-It accepts N future frames and delays decoded pictures by N-1 frames.
-The later frames are decoded in separate threads while the user is
-displaying the current one.
-
-Restrictions on clients
-==============================================
-
-Slice threading -
-* The client's draw_horiz_band() must be thread-safe according to the comment
-  in avcodec.h.
-
-Frame threading -
-* Restrictions with slice threading also apply.
-* For best performance, the client should set thread_safe_callbacks if it
-  provides a thread-safe get_buffer() callback.
-* There is one frame of delay added for every thread beyond the first one.
-  Clients must be able to handle this; the pkt_dts and pkt_pts fields in
-  AVFrame will work as usual.
-
-Restrictions on codec implementations
-==============================================
-
-Slice threading -
- None except that there must be something worth executing in parallel.
-
-Frame threading -
-* Codecs can only accept entire pictures per packet.
-* Codecs similar to ffv1, whose streams don't reset across frames,
-  will not work because their bitstreams cannot be decoded in parallel.
-
-* The contents of buffers must not be read before ff_thread_await_progress()
-  has been called on them. reget_buffer() and buffer age optimizations no longer work.
-* The contents of buffers must not be written to after ff_thread_report_progress()
-  has been called on them. This includes draw_edges().
-
-Porting codecs to frame threading
-==============================================
-
-Find all context variables that are needed by the next frame. Move all
-code changing them, as well as code calling get_buffer(), up to before
-the decode process starts. Call ff_thread_finish_setup() afterwards. If
-some code can't be moved, have update_thread_context() run it in the next
-thread.
-
-If the codec allocates writable tables in its init(), add an init_thread_copy()
-which re-allocates them for other threads.
-
-Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
-speed gain at this point but it should work.
-
-Call ff_thread_report_progress() after some part of the current picture has decoded.
-A good place to put this is where draw_horiz_band() is called - add this if it isn't
-called anywhere, as it's useful too and the implementation is trivial when you're
-doing this. Note that draw_edges() needs to be called before reporting progress.
-
-Before accessing a reference frame or its MVs, call ff_thread_await_progress().

doc/muxers.texi

@@ -1,738 +0,0 @@
-@chapter Muxers
-@c man begin MUXERS
-
-Muxers are configured elements in FFmpeg which allow writing
-multimedia streams to a particular type of file.
-
-When you configure your FFmpeg build, all the supported muxers
-are enabled by default. You can list all available muxers using the
-configure option @code{--list-muxers}.
-
-You can disable all the muxers with the configure option
-@code{--disable-muxers} and selectively enable / disable single muxers
-with the options @code{--enable-muxer=@var{MUXER}} /
-@code{--disable-muxer=@var{MUXER}}.
-
-The option @code{-formats} of the ff* tools will display the list of
-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
-
-CRC (Cyclic Redundancy Check) testing format.
-
-This muxer computes and prints the Adler-32 CRC 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
-CRC.
-
-The output of the muxer consists of a single line of the form:
-CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
-8 digits containing the CRC for all the decoded input frames.
-
-For example to compute the CRC of the input, and store it in the file
-@file{out.crc}:
-@example
-ffmpeg -i INPUT -f crc out.crc
-@end example
-
-You can print the CRC to stdout with the command:
-@example
-ffmpeg -i INPUT -f crc -
-@end example
-
-You can select the output format of each frame with @command{ffmpeg} by
-specifying the audio and video codec and format. For example to
-compute the CRC of 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 -
-@end example
-
-See also the @ref{framecrc} muxer.
-
-@anchor{framecrc}
-@section framecrc
-
-Per-packet 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
-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
-
-@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}:
-@example
-ffmpeg -i INPUT -f framecrc out.crc
-@end example
-
-To print the information to stdout, use 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
-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 -
-@end example
-
-See also the @ref{crc} muxer.
-
-@anchor{framemd5}
-@section framemd5
-
-Per-packet MD5 testing format.
-
-This muxer computes and prints the MD5 hash for each audio
-and video packet. By default audio frames are converted to signed
-16-bit raw audio and video frames to raw video before computing the
-hash.
-
-The output of the muxer consists of a line for each audio and video
-packet of the form:
-@example
-@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5}
-@end example
-
-@var{MD5} is a hexadecimal number representing the computed MD5 hash
-for the packet.
-
-For example to compute the MD5 of the audio and video frames in
-@file{INPUT}, converted to raw audio and video packets, and store it
-in the file @file{out.md5}:
-@example
-ffmpeg -i INPUT -f framemd5 out.md5
-@end example
-
-To print the information to stdout, use the command:
-@example
-ffmpeg -i INPUT -f framemd5 -
-@end example
-
-See also the @ref{md5} muxer.
-
-@anchor{hls}
-@section hls
-
-Apple HTTP Live Streaming muxer that segments MPEG-TS according to
-the HTTP Live Streaming specification.
-
-It creates a playlist file and numbered segment files. The output
-filename specifies the playlist filename; the segment filenames
-receive the same basename as the playlist, a sequential number and
-a .ts extension.
-
-@example
-ffmpeg -i in.nut out.m3u8
-@end example
-
-@table @option
-@item -hls_time @var{seconds}
-Set the segment length in seconds.
-@item -hls_list_size @var{size}
-Set the maximum number of playlist entries.
-@item -hls_wrap @var{wrap}
-Set the number after which index wraps.
-@item -start_number @var{number}
-Start the sequence from @var{number}.
-@end table
-
-@anchor{ico}
-@section ico
-
-ICO file muxer.
-
-Microsoft's icon file format (ICO) has some strict limitations that should be noted:
-
-@itemize
-@item
-Size cannot exceed 256 pixels in any dimension
-
-@item
-Only BMP and PNG images can be stored
-
-@item
-If a BMP image is used, it must be one of the following pixel formats:
-@example
-BMP Bit Depth      FFmpeg Pixel Format
-1bit               pal8
-4bit               pal8
-8bit               pal8
-16bit              rgb555le
-24bit              bgr24
-32bit              bgra
-@end example
-
-@item
-If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
-
-@item
-If a PNG image is used, it must use the rgba pixel format
-@end itemize
-
-@anchor{image2}
-@section image2
-
-Image file muxer.
-
-The image file muxer writes video frames to image files.
-
-The output filenames are specified by a pattern, which can be used to
-produce sequentially numbered series of files.
-The pattern may contain the string "%d" or "%0@var{N}d", this string
-specifies the position of the characters representing a numbering in
-the filenames. If the form "%0@var{N}d" is used, the string
-representing the number in each filename is 0-padded to @var{N}
-digits. The literal character '%' can be specified in the pattern with
-the string "%%".
-
-If the pattern contains "%d" or "%0@var{N}d", the first filename of
-the file list specified will contain the number 1, all the following
-numbers will be sequential.
-
-The pattern may contain a suffix which is used to automatically
-determine the format of the image files to write.
-
-For example the pattern "img-%03d.bmp" will specify a sequence of
-filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
-@file{img-010.bmp}, etc.
-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
-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'
-@end example
-
-Note that with @command{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'
-@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
-@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.
-
-This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
-
-The muxer options are:
-
-@table @option
-@item -mpegts_original_network_id @var{number}
-Set the original_network_id (default 0x0001). This is unique identifier
-of a network in DVB. Its main use is in the unique identification of a
-service through the path Original_Network_ID, Transport_Stream_ID.
-@item -mpegts_transport_stream_id @var{number}
-Set the transport_stream_id (default 0x0001). This identifies a
-transponder in DVB.
-@item -mpegts_service_id @var{number}
-Set the service_id (default 0x0001) also known as program in DVB.
-@item -mpegts_pmt_start_pid @var{number}
-Set the first PID for PMT (default 0x1000, max 0x1f00).
-@item -mpegts_start_pid @var{number}
-Set the first PID for data packets (default 0x0100, max 0x0f00).
-@end table
-
-The recognized metadata settings in mpegts muxer are @code{service_provider}
-and @code{service_name}. If they are not set the default for
-@code{service_provider} is "FFmpeg" and the default for
-@code{service_name} is "Service01".
-
-@example
-ffmpeg -i file.mpg -c copy \
-     -mpegts_original_network_id 0x1122 \
-     -mpegts_transport_stream_id 0x3344 \
-     -mpegts_service_id 0x5566 \
-     -mpegts_pmt_start_pid 0x1500 \
-     -mpegts_start_pid 0x150 \
-     -metadata service_provider="Some provider" \
-     -metadata service_name="Some Channel" \
-     -y out.ts
-@end example
-
-@section null
-
-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
-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}
-syntax.
-
-Alternatively you can write the command as:
-@example
-ffmpeg -benchmark -i INPUT -f null -
-@end example
-
-@section matroska
-
-Matroska container muxer.
-
-This muxer implements the matroska and webm container specs.
-
-The recognized metadata settings in this muxer are:
-
-@table @option
-
-@item title=@var{title name}
-Name provided to a single track
-@end table
-
-@table @option
-
-@item language=@var{language name}
-Specifies the language of the track in the Matroska languages form
-@end table
-
-@table @option
-
-@item stereo_mode=@var{mode}
-Stereo 3D video layout of two views in a single video track
-@table @option
-@item mono
-video is not stereo
-@item left_right
-Both views are arranged side by side, Left-eye view is on the left
-@item bottom_top
-Both views are arranged in top-bottom orientation, Left-eye view is at bottom
-@item top_bottom
-Both views are arranged in top-bottom orientation, Left-eye view is on top
-@item checkerboard_rl
-Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
-@item checkerboard_lr
-Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
-@item row_interleaved_rl
-Each view is constituted by a row based interleaving, Right-eye view is first row
-@item row_interleaved_lr
-Each view is constituted by a row based interleaving, Left-eye view is first row
-@item col_interleaved_rl
-Both views are arranged in a column based interleaving manner, Right-eye view is first column
-@item col_interleaved_lr
-Both views are arranged in a column based interleaving manner, Left-eye view is first column
-@item anaglyph_cyan_red
-All frames are in anaglyph format viewable through red-cyan filters
-@item right_left
-Both views are arranged side by side, Right-eye view is on the left
-@item anaglyph_green_magenta
-All frames are in anaglyph format viewable through green-magenta filters
-@item block_lr
-Both eyes laced in one Block, Left-eye view is first
-@item block_rl
-Both eyes laced in one Block, Right-eye view is first
-@end table
-@end table
-
-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
-@end example
-
-@c man end MUXERS

doc/nut.texi

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

doc/optimization.txt

@@ -17,15 +17,15 @@ Understanding these overoptimized functions:
 As many functions tend to be a bit difficult to understand because
 of optimizations, it can be hard to optimize them further, or write
 architecture-specific versions. It is recommended to look at older
-revisions of the interesting files (web frontends for the various FFmpeg
-branches are listed at http://ffmpeg.org/download.html).
+revisions of the interesting files (for a web frontend try ViewVC at
+http://svn.ffmpeg.org/ffmpeg/trunk/).
 Alternatively, look into the other architecture-specific versions in
 the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly
 comprehend the instructions, it could help understanding the functions
 and how they can be optimized.
 
 NOTE: If you still don't understand some function, ask at our mailing list!!!
-(http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel)
+(http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel)
 
 
 When is an optimization justified?
@@ -157,68 +157,15 @@ Use asm loops like:
 __asm__(
     "1: ....
     ...
-    "jump_instruction ....
+    "jump_instruciton ....
 Do not use C loops:
 do{
     __asm__(
         ...
 }while()
 
-For x86, mark registers that are clobbered in your asm. This means both
-general x86 registers (e.g. eax) as well as XMM registers. This last one is
-particularly important on Win64, where xmm6-15 are callee-save, and not
-restoring their contents leads to undefined results. In external asm (e.g.
-yasm), you do this by using:
-cglobal functon_name, num_args, num_regs, num_xmm_regs
-In inline asm, you specify clobbered registers at the end of your asm:
-__asm__(".." ::: "%eax").
-If gcc is not set to support sse (-msse) it will not accept xmm registers
-in the clobber list. For that we use two macros to declare the clobbers.
-XMM_CLOBBERS should be used when there are other clobbers, for example:
-__asm__(".." ::: XMM_CLOBBERS("xmm0",) "eax");
-and XMM_CLOBBERS_ONLY should be used when the only clobbers are xmm registers:
-__asm__(".." :: XMM_CLOBBERS_ONLY("xmm0"));
-
-Do not expect a compiler to maintain values in your registers between separate
-(inline) asm code blocks. It is not required to. For example, this is bad:
-__asm__("movdqa %0, %%xmm7" : src);
-/* do something */
-__asm__("movdqa %%xmm7, %1" : dst);
-- first of all, you're assuming that the compiler will not use xmm7 in
-   between the two asm blocks.  It probably won't when you test it, but it's
-   a poor assumption that will break at some point for some --cpu compiler flag
-- secondly, you didn't mark xmm7 as clobbered. If you did, the compiler would
-   have restored the original value of xmm7 after the first asm block, thus
-   rendering the combination of the two blocks of code invalid
-Code that depends on data in registries being untouched, should be written as
-a single __asm__() statement. Ideally, a single function contains only one
-__asm__() block.
-
-Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics.
-The latter requires a good optimizing compiler which gcc is not.
-
-Inline asm vs. external asm
----------------------------
-Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)
-and external asm (.s or .asm files, handled by an assembler such as yasm/nasm)
-are accepted in FFmpeg. Which one to use differs per specific case.
-
-- if your code is intended to be inlined in a C function, inline asm is always
-   better, because external asm cannot be inlined
-- if your code calls external functions, yasm is always better
-- if your code takes huge and complex structs as function arguments (e.g.
-   MpegEncContext; note that this is not ideal and is discouraged if there
-   are alternatives), then inline asm is always better, because predicting
-   member offsets in complex structs is almost impossible. It's safest to let
-   the compiler take care of that
-- in many cases, both can be used and it just depends on the preference of the
-   person writing the asm. For new asm, the choice is up to you. For existing
-   asm, you'll likely want to maintain whatever form it is currently in unless
-   there is a good reason to change it.
-- if, for some reason, you believe that a particular chunk of existing external
-   asm could be improved upon further if written in inline asm (or the other
-   way around), then please make the move from external asm <-> inline asm a
-   separate patch before your patches that actually improve the asm.
+Use __asm__() instead of intrinsics. The latter requires a good optimizing compiler
+which gcc is not.
 
 
 Links:
@@ -253,7 +200,7 @@ Optimization guide for ARM11 (used in Nokia N800 Internet Tablet):
 http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf
 Optimization guide for Intel XScale (used in Sharp Zaurus PDA):
 http://download.intel.com/design/intelxscale/27347302.pdf
-Intel Wireless MMX 2 Coprocessor: Programmers Reference Manual
+Intel Wireless MMX2 Coprocessor: Programmers Reference Manual
 http://download.intel.com/design/intelxscale/31451001.pdf
 
 PowerPC-specific:

doc/outdevs.texi

@@ -1,156 +0,0 @@
-@chapter Output Devices
-@c man begin OUTPUT DEVICES
-
-Output devices are configured elements in FFmpeg which allow to write
-multimedia data to an output device attached to your system.
-
-When you configure your FFmpeg build, all the supported output devices
-are enabled by default. You can list all available ones using the
-configure option "--list-outdevs".
-
-You can disable all the output devices using the configure option
-"--disable-outdevs", and selectively enable an output device using the
-option "--enable-outdev=@var{OUTDEV}", or you can disable a particular
-input device using the option "--disable-outdev=@var{OUTDEV}".
-
-The option "-formats" of the ff* tools will display the list of
-enabled output devices (amongst the muxers).
-
-A description of the currently available output devices follows.
-
-@section alsa
-
-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.
-
-This output devices allows to show a video stream in an SDL
-window. Only one SDL 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 libsdl installed on your system
-when configuring your build.
-
-For more information about SDL, check:
-@url{http://www.libsdl.org/}
-
-@subsection Options
-
-@table @option
-
-@item window_title
-Set the SDL window title, if not specified default to the filename
-specified for the output device.
-
-@item icon_title
-Set the name of the iconified SDL window, if not specified it is set
-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.
-@end table
-
-@subsection Examples
-
-The following command shows the @command{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"
-@end example
-
-@section sndio
-
-sndio audio output device.
-
-@c man end OUTPUT DEVICES