Update back for git master

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Changelog: replace next by 2.0
2013-07-10 03:14:23 +02:00 · 2013-07-10 02:47:10 +02:00 · 2013-07-10 02:15:43 +02:00 · 2013-07-10 02:15:43 +02:00 · 2013-07-10 01:18:59 +02:00 · 2013-07-10 00:57:04 +02:00
2687 changed files with 205706 additions and 115659 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -3,10 +3,16 @@
 *.d
 *.def
 *.dll
+*.dylib
 *.exe
+*.exp
+*.gcda
+*.gcno
 *.h.c
+*.ilk
 *.lib
 *.pc
+*.pdb
 *.so
 *.so.*
 *.ver
@@ -20,10 +26,13 @@
 /ffprobe
 /ffserver
 /config.*
+/coverage.info
 /version.h
 /doc/*.1
+/doc/*.3
 /doc/*.html
 /doc/*.pod
+/doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/examples/decoding_encoding
@@ -32,10 +41,13 @@
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
+/doc/examples/pc-uninstalled
+/doc/examples/resampling_audio
 /doc/examples/scaling_video
 /doc/fate.txt
+/doc/doxy/html/
 /doc/print_options
-/doxy/
+/lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
@@ -45,16 +57,22 @@
 /tests/data/
 /tests/rotozoom
 /tests/tiny_psnr
+/tests/tiny_ssim
 /tests/videogen
 /tests/vsynth1/
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
 /tools/cws2fws
+/tools/fourcc2pixfmt
+/tools/ffescape
 /tools/ffeval
+/tools/ffhash
 /tools/graph2dot
 /tools/ismindex
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
 /tools/trasher
+/tools/seek_print
+/tools/zmqsend
--- a/59
+++ b/59
@@ -1,55 +1,6 @@
-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-currently quite outdated and git serves as a much better tool for determining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.
+See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
+get the names of people who have contributed to FFmpeg.

-Dénes Balatoni
-Michel Bardiaux
-Fabrice Bellard
-Patrice Bensoussan
-Alex Beregszaszi
-BERO
-Thilo Borgmann
-Mario Brito
-Ronald Bultje
-Alex Converse
-Maarten Daniels
-Reimar Doeffinger
-Tim Ferguson
-Brian Foley
-Arpad Gereoffy
-Philip Gladstone
-Vladimir Gneushev
-Roine Gustafsson
-David Hammerton
-Wolfgang Hesseler
-Marc Hoffman
-Falk Hueffner
-Aurélien Jacobs
-Steven Johnson
-Zdenek Kabelac
-Robin Kay
-Todd Kirby
-Nick Kurshev
-Benjamin Larsson
-Loïc Le Loarer
-Daniel Maas
-Mike Melanson
-Loren Merritt
-Jeff Muizelaar
-Michael Niedermayer
-François Revol
-Peter Ross
-Måns Rullgård
-Stefano Sabatini
-Roman Shaposhnik
-Oded Shimon
-Dieter Shirley
-Konstantin Shishkov
-Juan J. Sierralta
-Ewald Snel
-Sascha Sommer
-Leon van Stuivenberg
-Roberto Togni
-Lionel Ulmer
-Reynaldo Verdejo
+To check the log, you can type the command "git log" in the FFmpeg
+source directory, or browse the online repository at
+http://source.ffmpeg.org.
--- a/162
+++ b/162
@@ -1,10 +1,166 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

-version next:
+version 2.0:
+
+- curves filter
+- reference-counting for AVFrame and AVPacket data
+- ffmpeg now fails when input options are used for output file
+  or vice versa
+- support for Monkey's Audio versions from 3.93
+- perms and aperms filters
+- audio filtering support in ffplay
+- 10% faster aac encoding on x86 and MIPS
+- sine audio filter source
+- WebP demuxing and decoding support
+- new ffmpeg options -filter_script and -filter_complex_script, which allow a
+  filtergraph description to be read from a file
+- OpenCL support
+- audio phaser filter
+- separatefields filter
+- libquvi demuxer
+- uniform options syntax across all filters
+- telecine filter
+- new interlace filter
+- smptehdbars source
+- inverse telecine filters (fieldmatch and decimate)
+- colorbalance filter
+- colorchannelmixer filter
+- The matroska demuxer can now output proper verbatim ASS packets. It will
+  become the default at the next libavformat major bump.
+- decent native animated GIF encoding
+- asetrate filter
+- interleave filter
+- timeline editing with filters
+- vidstabdetect and vidstabtransform filters for video stabilization using
+  the vid.stab library
+- astats filter
+- trim and atrim filters
+- ffmpeg -t and -ss (output-only) options are now sample-accurate when
+  transcoding audio
+- Matroska muxer can now put the index at the beginning of the file.
+- extractplanes filter
+- avectorscope filter
+- ADPCM DTK decoder
+- ADP demuxer
+- RSD demuxer
+- RedSpark demuxer
+- ADPCM IMA Radical decoder
+- zmq filters
+- DCT denoiser filter (dctdnoiz)
+- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
+- Apple Intermediate Codec decoder
+- Escape 130 video decoder
+- FTP protocol support
+- V4L2 output device
+- 3D LUT filter (lut3d)
+- SMPTE 302M audio encoder
+- support for slice multithreading in libavfilter
+- Hald CLUT support (generation and filtering)
+- VC-1 interlaced B-frame support
+- support for WavPack muxing (raw and in Matroska)
+- XVideo output device
+- vignette filter
+- True Audio (TTA) encoder
+- Go2Webinar decoder
+- mcdeint filter ported from libmpcodecs
+- sab filter ported from libmpcodecs
+- ffprobe -show_chapters option
+- WavPack encoding through libwavpack
+- rotate filter
+- spp filter ported from libmpcodecs
+- libgme support
+- psnr filter
+
+
+version 1.2:
+
+- VDPAU hardware acceleration through normal hwaccel
+- SRTP support
+- Error diffusion dither in Swscale
+- Chained Ogg support
+- Theora Midstream reconfiguration support
+- EVRC decoder
+- audio fade filter
+- filtering audio with unknown channel layout
+- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
+  and treble audio filter
+- improved showspectrum filter, with multichannel support and sox-like colors
+- histogram filter
+- tee muxer
+- il filter ported from libmpcodecs
+- support ID3v2 tags in ASF files
+- encrypted TTA stream decoding support
+- RF64 support in WAV muxer
+- noise filter ported from libmpcodecs
+- Subtitles character encoding conversion
+- blend filter
+- stereo3d filter ported from libmpcodecs
+
+
+version 1.1:
+
+- stream disposition information printing in ffprobe
+- filter for loudness analysis following EBU R128
+- Opus encoder using libopus
+- ffprobe -select_streams option
+- Pinnacle TARGA CineWave YUV16 decoder
+- TAK demuxer, decoder and parser
+- DTS-HD demuxer
+- remove -same_quant, it hasn't worked for years
+- FFM2 support
+- X-Face image encoder and decoder
+- 24-bit FLAC encoding
+- multi-channel ALAC encoding up to 7.1
+- metadata (INFO tag) support in WAV muxer
+- subtitles raw text decoder
+- support for building DLLs using MSVC
+- LVF demuxer
+- ffescape tool
+- metadata (info chunk) support in CAF muxer
+- field filter ported from libmpcodecs
+- AVR demuxer
+- geq filter ported from libmpcodecs
+- remove ffserver daemon mode
+- AST muxer/demuxer
+- new expansion syntax for drawtext
+- BRender PIX image decoder
+- ffprobe -show_entries option
+- ffprobe -sections option
+- ADPCM IMA Dialogic decoder
+- BRSTM demuxer
+- animated GIF decoder and demuxer
+- PVF demuxer
+- subtitles filter
+- IRCAM muxer/demuxer
+- Paris Audio File demuxer
+- Virtual concatenation demuxer
+- VobSub demuxer
+- JSON captions for TED talks decoding support
+- SOX Resampler support in libswresample
+- aselect filter
+- SGI RLE 8-bit decoder
+- Silicon Graphics Motion Video Compressor 1 & 2 decoder
+- Silicon Graphics Movie demuxer
+- apad filter
+- Resolution & pixel format change support with multithreading for H.264
+- documentation split into per-component manuals
+- pp (postproc) filter ported from MPlayer
+- NIST Sphere demuxer
+- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders
+- Sony Wave64 muxer
+- adobe and limelight publisher authentication in RTMP
+- data: URI scheme
+- support building on the Plan 9 operating system
+- kerndeint filter ported from MPlayer
+- histeq filter ported from VirtualDub
+- Megalux Frame demuxer
+- 012v decoder
+- Improved AVC Intra decoding support


 version 1.0:
+
 - INI and flat output in ffprobe
 - Scene detection in libavfilter
 - Indeo Audio decoder
@@ -57,8 +213,6 @@ version 1.0:
 - volume measurement filter
 - Ut Video encoder
 - Microsoft Screen 2 decoder
- Matroska demuxer now identifies SRT subtitles as AV_CODEC_ID_SUBRIP
-  instead of AV_CODEC_ID_TEXT
 - smartblur filter ported from MPlayer
 - CPiA decoder
 - decimate filter ported from MPlayer
@@ -69,6 +223,7 @@ version 1.0:
 - WebVTT demuxer and decoder (simple tags supported)
 - RTP packetization of JPEG
 - faststart option in the MOV/MP4 muxer
+- support for building with MSVC


 version 0.11:
@@ -112,6 +267,7 @@ version 0.11:
 - vorbis parser
 - png parser
 - audio mix filter
+- ffv1: support (draft) version 1.3


 version 0.10:
--- a/35
+++ b/35
@@ -23,18 +23,37 @@ Specifically, the GPL parts of FFmpeg are
  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_mcdeint.c
    - vf_mp.c
+    - vf_noise.c
+    - vf_owdenoise.c
+    - vf_pp.c
+    - vf_sab.c
+    - vf_smartblur.c
+    - vf_spp.c
+    - vf_stereo3d.c
    - vf_super2xsai.c
    - vf_tinterlace.c
    - vf_yadif.c
    - vsrc_mptestsrc.c

+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
+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.
+
 There are a handful of files under other licensing terms, namely:

 * The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
@@ -44,11 +63,6 @@ There are a handful of files under other licensing terms, namely:
  You must also indicate any changes including additions and deletions to
  those three files in the documentation.

-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
-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
 ==================
@@ -59,8 +73,15 @@ affect the licensing of binaries resulting from the combination.
 compatible libraries
 --------------------

-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
+The following libraries are under GPL:
+    - frei0r
+    - libcdio
+    - libutvideo
+    - libvidstab
+    - libx264
+    - libxavs
+    - libxvid
+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
--- a/73
+++ b/73
@@ -7,8 +7,8 @@ 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.
+A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
+patches and related discussions.


 Project Leader
@@ -67,6 +67,8 @@ Other:
  mathematics.c, mathematics.h          Michael Niedermayer
  integer.c, integer.h                  Michael Niedermayer
  bswap.h
+  opencl.c, opencl.h                    Wei Gao
+  ripemd.c, ripemd.h                    James Almer


 libavcodec
@@ -130,6 +132,7 @@ 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
@@ -162,6 +165,7 @@ Codecs:
  interplayvideo.c                      Mike Melanson
  ivi*                                  Kostya Shishkov
  jacosub*                              Clément Bœsch
+  jpeg2000*                             Nicolas Bertrand
  jpeg_ls.c                             Kostya Shishkov
  jvdec.c                               Peter Ross
  kmvc.c                                Kostya Shishkov
@@ -182,17 +186,19 @@ Codecs:
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
  mimic.c                               Ramiro Polla
-  mjpeg.c                               Michael Niedermayer
+  mjpeg*.c                              Michael Niedermayer
  mlp*                                  Ramiro Polla
  mmvideo.c                             Peter Ross
  mpc*                                  Kostya Shishkov
  mpeg12.c, mpeg12data.h                Michael Niedermayer
  mpegvideo.c, mpegvideo.h              Michael Niedermayer
+  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
  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
@@ -212,11 +218,13 @@ Codecs:
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
+  smvjpegdec.c                          Ash Hughes
  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
@@ -224,6 +232,7 @@ Codecs:
  truespeech.c                          Kostya Shishkov
  tscc.c                                Kostya Shishkov
  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
+  ttaenc.c                              Paul B Mahol
  txd.c                                 Ivo van Poorten
  ulti*                                 Kostya Shishkov
  v410*.c                               Derek Buitenhuis
@@ -232,6 +241,7 @@ Codecs:
  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
@@ -246,8 +256,10 @@ 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

@@ -270,7 +282,7 @@ libavdevice
  libdc1394.c                           Roman Shaposhnik
  v4l2.c                                Luca Abeni
  vfwcap.c                              Ramiro Polla
-
+  dshow.c                               Roger Pack

 libavfilter
 ===========
@@ -283,6 +295,8 @@ Filters:
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
  af_pan.c                              Nicolas George
+  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
+  vf_drawbox.c/drawgrid                 Andrey Utkin
  vf_yadif.c                            Michael Niedermayer

 Sources:
@@ -301,17 +315,26 @@ 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
+  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
+  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
@@ -322,19 +345,22 @@ Muxers/Demuxers:
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
  ipmovie.c                             Mike Melanson
-  img2.c                                Michael Niedermayer
+  img2*.c                               Michael Niedermayer
+  ircam*                                Paul B Mahol
  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
@@ -346,6 +372,7 @@ Muxers/Demuxers:
  mtv.c                                 Reynaldo H. Verdejo Pinochet
  mxf*                                  Baptiste Coudurier
  mxfdec.c                              Tomas Härdin
+  nistspheredec.c                       Paul B Mahol
  nsvdec.c                              Francois Revol
  nut.c                                 Michael Niedermayer
  nuv.c                                 Reimar Doeffinger
@@ -353,8 +380,10 @@ Muxers/Demuxers:
  oggenc.c                              Baptiste Coudurier
  oggparse*.c                           David Conrad
  oma.c                                 Maxim Poliakovski
+  paf.c                                 Paul B Mahol
  psxstr.c                              Mike Melanson
  pva.c                                 Ivo van Poorten
+  pvfdec.c                              Paul B Mahol
  r3d.c                                 Baptiste Coudurier
  raw.c                                 Michael Niedermayer
  rdt.c                                 Ronald S. Bultje
@@ -370,24 +399,43 @@ Muxers/Demuxers:
  segafilm.c                            Mike Melanson
  siff.c                                Kostya Shishkov
  smacker.c                             Kostya Shishkov
+  smjpeg*                               Paul B Mahol
  srtdec.c                              Aurelien Jacobs
  swf.c                                 Baptiste Coudurier
+  takdec.c                              Paul B Mahol
  tta.c                                 Alex Beregszaszi
  txd.c                                 Ivo van Poorten
  voc.c                                 Aurelien Jacobs
  wav.c                                 Michael Niedermayer
  wc3movie.c                            Mike Melanson
+  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
  wv.c                                  Kostya Shishkov
+  wvenc.c                               Paul B Mahol

 Protocols:
  bluray.c                              Petri Hintukainen
+  ftp.c                                 Lukasz Marek
  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
 =====================================

@@ -408,10 +456,11 @@ x86                                     Michael Niedermayer
 Releases
 ========

-1.0                                     Michael Niedermayer
-0.11                                    Michael Niedermayer
-0.10                                    Michael Niedermayer
+2.0                                     Michael Niedermayer
+1.2                                     Michael Niedermayer
+1.1                                     Michael Niedermayer

+If you want to maintain an older release, please contact us


 GnuPG Fingerprints of maintainers and contributors
@@ -419,6 +468,7 @@ 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
+Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
 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
@@ -426,8 +476,10 @@ 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
+FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
+Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 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
@@ -441,6 +493,7 @@ 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              9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
+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                  D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD
+Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
--- a/42
+++ b/42
@@ -17,9 +17,10 @@ PROGS-$(CONFIG_FFSERVER) += ffserver

 PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-OBJS        = cmdutils.o
+
+OBJS        = cmdutils.o $(EXEOBJS)
 OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
@@ -27,7 +28,6 @@ TOOLS-$(CONFIG_ZLIB) += cws2fws
 BASENAMES   = ffmpeg ffplay ffprobe ffserver
 ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-ALLMANPAGES = $(BASENAMES:%=%.1)

 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
@@ -41,9 +41,9 @@ 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
+EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README

-SKIPHEADERS = cmdutils_common_opts.h
+SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h

 include $(SRC_PATH)/common.mak

@@ -56,8 +56,8 @@ $(PROGS): %$(EXESUF): %_g$(EXESUF)
 	$(CP) $< $@
 	$(STRIP) $@

-$(TOOLS): %$(EXESUF): %.o
-	$(LD) $(LDFLAGS) $(LD_O) $< $(ELIBS)
+$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
+	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)

 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)

@@ -68,9 +68,9 @@ config.h: .config
 	@-tput sgr0 2>/dev/null

 SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
-               ARCH_HEADERS BUILT_HEADERS SKIPHEADERS                    \
-               ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS NEON-OBJS             \
-               MMI-OBJS ALTIVEC-OBJS VIS-OBJS                            \
+               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
@@ -91,8 +91,8 @@ endef
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))

 define DOPROG
-OBJS-$(1) += $(1).o
-$(1)$(PROGSSUF)_g$(EXESUF): $(OBJS-$(1))
+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))
@@ -101,8 +101,8 @@ endef

 $(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))

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

 OBJDIRS += tools

@@ -152,27 +152,17 @@ uninstall-data:
 clean::
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(TOOLS)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
-	$(RM) coverage.info
 	$(RM) -r coverage-html
+	$(RM) -rf coverage.info lcov

 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .version version.h libavutil/avconfig.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.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' > $@
-
-coverage-html: coverage.info
-	$(Q)mkdir -p $@
-	$(Q)genhtml -o $@ $<
-	$(Q)touch $@
-
 check: all alltools examples testprogs fate

 include $(SRC_PATH)/doc/Makefile
--- a/2
+++ b/2
@@ -1 +1 @@
-1.0.git
+2.0
--- a/arch.mak
+++ b/arch.mak
@@ -1,9 +1,8 @@
 OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
 OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_ARMVFP)  += $(ARMVFP-OBJS)  $(ARMVFP-OBJS-yes)
+OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)

-OBJS-$(HAVE_MMI)     += $(MMI-OBJS)     $(MMI-OBJS-yes)
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
--- a/cmdutils.c
+++ b/cmdutils.c
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -51,8 +51,18 @@ extern const int this_year;
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
-extern struct SwrContext *swr_opts;
-extern AVDictionary *format_opts, *codec_opts;
+extern AVDictionary *swr_opts;
+extern AVDictionary *format_opts, *codec_opts, *resample_opts;
+
+/**
+ * Register a program-specific cleanup routine.
+ */
+void register_exit(void (*cb)(int ret));
+
+/**
+ * Wraps exit with a program-specific cleanup routine.
+ */
+void exit_program(int ret);

 /**
 * Initialize the cmdutils option system, in particular
@@ -90,6 +100,8 @@ 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_opencl(void *optctx, const char *opt, const char *arg);
+
 /**
 * Limit the execution time.
 */
@@ -123,7 +135,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
 * not zero timestr is interpreted as a duration, otherwise as a
 * date
 *
- * @see parse_date()
+ * @see av_parse_time()
 */
 int64_t parse_time_or_die(const char *context, const char *timestr,
                          int is_duration);
@@ -139,7 +151,7 @@ typedef struct SpecifierOpt {
    } u;
 } SpecifierOpt;

-typedef struct {
+typedef struct OptionDef {
    const char *name;
    int flags;
 #define HAS_ARG    0x0001
@@ -162,6 +174,8 @@ typedef struct {
                                   an int containing element count in the array. */
 #define OPT_TIME  0x10000
 #define OPT_DOUBLE 0x20000
+#define OPT_INPUT  0x40000
+#define OPT_OUTPUT 0x80000
     union {
        void *dst_ptr;
        int (*func_arg)(void *, const char *, const char *);
@@ -190,13 +204,13 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
 void show_help_children(const AVClass *class, int flags);

 /**
- * Per-avtool specific help handler. Implemented in each
- * avtool, called by show_help().
+ * 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 avtools.
+ * Generic -h handler common to all fftools.
 */
 int show_help(void *optctx, const char *opt, const char *arg);

@@ -204,6 +218,8 @@ int show_help(void *optctx, const char *opt, const char *arg);
 * Parse the command line arguments.
 *
 * @param optctx an opaque options context
+ * @param argc   number of command line arguments
+ * @param argv   values of command line arguments
 * @param options Array with the definitions required to interpret every
 * option of the form: -option_name [argument]
 * @param parse_arg_function Name of the function called to process every
@@ -221,6 +237,101 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
 int parse_option(void *optctx, const char *opt, const char *arg,
                 const OptionDef *options);

+/**
+ * An option extracted from the commandline.
+ * Cannot use AVDictionary because of options like -map which can be
+ * used multiple times.
+ */
+typedef struct Option {
+    const OptionDef  *opt;
+    const char       *key;
+    const char       *val;
+} Option;
+
+typedef struct OptionGroupDef {
+    /**< group name */
+    const char *name;
+    /**
+     * Option to be used as group separator. Can be NULL for groups which
+     * are terminated by a non-option argument (e.g. ffmpeg output files)
+     */
+    const char *sep;
+    /**
+     * Option flags that must be set on each option that is
+     * applied to this group
+     */
+    int flags;
+} OptionGroupDef;
+
+typedef struct OptionGroup {
+    const OptionGroupDef *group_def;
+    const char *arg;
+
+    Option *opts;
+    int  nb_opts;
+
+    AVDictionary *codec_opts;
+    AVDictionary *format_opts;
+    AVDictionary *resample_opts;
+    struct SwsContext *sws_opts;
+    AVDictionary *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.
 */
@@ -249,6 +360,8 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
 * Create a new options dictionary containing only the options from
 * opts which apply to the codec with ID codec_id.
 *
+ * @param opts     dictionary to place options in
+ * @param codec_id ID of the codec that should be filtered for
 * @param s Corresponding format context.
 * @param st A stream from s for which the options should be filtered.
 * @param codec The particular codec for which the options should be filtered.
@@ -382,6 +495,7 @@ int read_yesno(void);
 * Read the file with name filename, and put its content in a newly
 * allocated 0-terminated buffer.
 *
+ * @param filename file to read from
 * @param bufptr location where pointer to buffer is returned
 * @param size   location where size of buffer is returned
 * @return 0 in case of success, a negative value corresponding to an
@@ -410,64 +524,22 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
 FILE *get_preset_file(char *filename, size_t filename_size,
                      const char *preset_name, int is_path, const char *codec_name);

-/**
- * Do all the necessary cleanup and abort.
- * This function is implemented in the avtools, not cmdutils.
- */
-av_noreturn void exit_program(int ret);
-
 /**
 * Realloc array to hold new_size elements of elem_size.
- * Calls exit_program() on failure.
+ * 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);

-typedef struct FrameBuffer {
-    uint8_t *base[4];
-    uint8_t *data[4];
-    int  linesize[4];
+#define media_type_string av_get_media_type_string

-    int h, w;
-    enum PixelFormat 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 GROW_ARRAY(array, nb_elems)\
+    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)

 #define GET_PIX_FMT_NAME(pix_fmt)\
    const char *name = av_get_pix_fmt_name(pix_fmt);
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -14,10 +14,11 @@
    { "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" },
-    { "debug"      , HAS_ARG,  {.func_arg = opt_codec_debug},   "set debug flags", "flags" },
-    { "fdebug"     , HAS_ARG,  {.func_arg = opt_codec_debug},   "set debug flags", "flags" },
+    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
+    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set 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" },
+#if CONFIG_OPENCL
+    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
+#endif
--- a/common.mak
+++ b/common.mak
@@ -30,9 +30,9 @@ CFLAGS     += $(ECFLAGS)
 CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
-YASMFLAGS  += $(IFLAGS:%=%/) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
+YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm

-HOSTCCFLAGS = $(IFLAGS) $(HOSTCFLAGS)
+HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)

 define COMPILE
@@ -56,6 +56,9 @@ COMPILE_S = $(call COMPILE,AS)
 %.o: %.S
 	$(COMPILE_S)

+%.i: %.c
+	$(CC) $(CCFLAGS) $(CC_E) $<
+
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@

@@ -94,9 +97,11 @@ HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
 TOOLS     += $(TOOLS-yes)
 TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
+HEADERS   += $(HEADERS-yes)

 DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))

+SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
@@ -113,18 +118,24 @@ $(HOSTPROGS): %$(HOSTEXESUF): %.o
 	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)

 $(OBJS):     | $(sort $(dir $(OBJS)))
+$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
-$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(TOOLOBJS): | tools

-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))

 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a

+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))
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -0,0 +1,14 @@
+/*
+ * Workaround aix-specific class() function clashing with ffmpeg class usage
+ */
+
+#ifndef COMPAT_AIX_MATH_H
+#define COMPAT_AIX_MATH_H
+
+#define class class_in_math_h_causes_problems
+
+#include_next <math.h>
+
+#undef class
+
+#endif /* COMPAT_AIX_MATH_H */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -0,0 +1,879 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+// NOTE: this is a partial update of the Avisynth C interface to recognize
+// new color spaces added in Avisynth 2.60. By no means is this document
+// completely Avisynth 2.60 compliant.
+
+#ifndef __AVISYNTH_C__
+#define __AVISYNTH_C__
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#define AVSC_USE_STDCALL 1
+
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
+#endif
+
+#define AVSC_INLINE static __inline
+
+#ifdef AVISYNTH_C_EXPORTS
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+typedef unsigned char BYTE;
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// Constants
+//
+
+#ifndef __AVISYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 4 };
+#endif
+
+enum {AVS_SAMPLE_INT8  = 1<<0,
+      AVS_SAMPLE_INT16 = 1<<1,
+      AVS_SAMPLE_INT24 = 1<<2,
+      AVS_SAMPLE_INT32 = 1<<3,
+      AVS_SAMPLE_FLOAT = 1<<4};
+
+enum {AVS_PLANAR_Y=1<<0,
+      AVS_PLANAR_U=1<<1,
+      AVS_PLANAR_V=1<<2,
+      AVS_PLANAR_ALIGNED=1<<3,
+      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_A=1<<4,
+      AVS_PLANAR_R=1<<5,
+      AVS_PLANAR_G=1<<6,
+      AVS_PLANAR_B=1<<7,
+      AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
+
+  // Colorspace properties.
+enum {AVS_CS_BGR = 1<<28,
+      AVS_CS_YUV = 1<<29,
+      AVS_CS_INTERLEAVED = 1<<30,
+      AVS_CS_PLANAR = 1<<31,
+
+      AVS_CS_SHIFT_SUB_WIDTH   = 0,
+      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
+      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
+
+      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
+      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
+      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
+      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
+
+      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
+      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
+
+      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
+      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
+      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
+      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
+
+      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
+
+      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
+      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
+
+  // Specific colorformats
+enum {
+  AVS_CS_UNKNOWN = 0,
+  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
+  //  AVS_CS_YV12  = 1<<3  Reserved
+  //  AVS_CS_I420  = 1<<4  Reserved
+  AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
+
+  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
+  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
+  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:0 planar
+  AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
+  AVS_CS_IYUV  = AVS_CS_I420,
+  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
+  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
+  AVS_CS_Y8    = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
+};
+
+enum {
+  AVS_IT_BFF = 1<<0,
+  AVS_IT_TFF = 1<<1,
+  AVS_IT_FIELDBASED = 1<<2};
+
+enum {
+  AVS_FILTER_TYPE=1,
+  AVS_FILTER_INPUT_COLORSPACE=2,
+  AVS_FILTER_OUTPUT_TYPE=9,
+  AVS_FILTER_NAME=4,
+  AVS_FILTER_AUTHOR=5,
+  AVS_FILTER_VERSION=6,
+  AVS_FILTER_ARGS=7,
+  AVS_FILTER_ARGS_INFO=8,
+  AVS_FILTER_ARGS_DESCRIPTION=10,
+  AVS_FILTER_DESCRIPTION=11};
+
+enum {  //SUBTYPES
+  AVS_FILTER_TYPE_AUDIO=1,
+  AVS_FILTER_TYPE_VIDEO=2,
+  AVS_FILTER_OUTPUT_TYPE_SAME=3,
+  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
+
+enum {
+  AVS_CACHE_NOTHING=0,
+  AVS_CACHE_RANGE=1,
+  AVS_CACHE_ALL=2,
+  AVS_CACHE_AUDIO=3,
+  AVS_CACHE_AUDIO_NONE=4,
+  AVS_CACHE_AUDIO_AUTO=5
+  };
+
+#define AVS_FRAME_ALIGN 16
+
+typedef struct AVS_Clip AVS_Clip;
+typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoInfo
+//
+
+// AVS_VideoInfo is layed out identicly to VideoInfo
+typedef struct AVS_VideoInfo {
+  int width, height;    // width=0 means no video
+  unsigned fps_numerator, fps_denominator;
+  int num_frames;
+
+  int pixel_type;
+
+  int audio_samples_per_second;   // 0 means no audio
+  int sample_type;
+  INT64 num_audio_samples;
+  int nchannels;
+
+  // Imagetype properties
+
+  int image_type;
+} AVS_VideoInfo;
+
+// useful functions of the above
+AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
+        { return (p->width!=0); }
+
+AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
+        { return (p->audio_samples_per_second!=0); }
+
+AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_BGR); }
+
+AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
+        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
+
+AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
+
+AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_YUV ); }
+
+AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
+
+AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
+
+AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
+
+AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
+
+AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
+
+AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
+
+AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
+        { return ((p->pixel_type & property)==property ); }
+
+AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type & AVS_CS_PLANAR); }
+
+AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
+        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
+
+AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_FIELDBASED); }
+
+AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
+        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
+
+AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_BFF); }
+
+AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_TFF); }
+
+AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
+{
+  switch (p->pixel_type) {
+      case AVS_CS_BGR24: return 24;
+      case AVS_CS_BGR32: return 32;
+      case AVS_CS_YUY2:  return 16;
+      case AVS_CS_YV12:
+      case AVS_CS_I420:  return 12;
+      default:           return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
+        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
+
+AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
+        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
+
+AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
+        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
+
+AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
+        { return p->audio_samples_per_second; }
+
+
+AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
+{
+    switch (p->sample_type) {
+      case AVS_SAMPLE_INT8:  return sizeof(signed char);
+      case AVS_SAMPLE_INT16: return sizeof(signed short);
+      case AVS_SAMPLE_INT24: return 3;
+      case AVS_SAMPLE_INT32: return sizeof(signed int);
+      case AVS_SAMPLE_FLOAT: return sizeof(float);
+      default: return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
+        { return p->nchannels*avs_bytes_per_channel_sample(p);}
+
+AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
+        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
+
+AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
+
+AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
+        { return bytes / avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return samples * avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
+        { return p->nchannels; }
+
+AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
+        { return p->sample_type;}
+
+// useful mutator
+AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
+        { p->image_type|=property; }
+
+AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
+        { p->image_type&=~property; }
+
+AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
+        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
+
+AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
+{
+    unsigned x=numerator, y=denominator;
+    while (y) {   // find gcd
+      unsigned t = x%y; x = y; y = t;
+    }
+    p->fps_numerator = numerator/x;
+    p->fps_denominator = denominator/x;
+}
+
+AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
+{
+        return (x->pixel_type == y->pixel_type)
+                || (avs_is_yv12(x) && avs_is_yv12(y));
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoFrame
+//
+
+// VideoFrameBuffer holds information about a memory block which is used
+// for video data.  For efficiency, instances of this class are not deleted
+// when the refcount reaches zero; instead they're stored in a linked list
+// to be reused.  The instances are deleted when the corresponding AVS
+// file is closed.
+
+// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrameBuffer {
+  BYTE * data;
+  int data_size;
+  // sequence_number is incremented every time the buffer is changed, so
+  // that stale views can tell they're no longer valid.
+  volatile long sequence_number;
+
+  volatile long refcount;
+} AVS_VideoFrameBuffer;
+
+// VideoFrame holds a "window" into a VideoFrameBuffer.
+
+// AVS_VideoFrame is layed out identicly to IVideoFrame
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrame {
+  volatile long refcount;
+  AVS_VideoFrameBuffer * vfb;
+  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
+  int row_sizeUV, heightUV;
+} AVS_VideoFrame;
+
+// Access functions for AVS_VideoFrame
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return p->pitch;}
+
+AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
+  switch (plane) {
+  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
+  return p->pitch;}
+
+AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
+        return p->row_size; }
+
+AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
+        int r;
+    switch (plane) {
+    case AVS_PLANAR_U: case AVS_PLANAR_V:
+                if (p->pitchUV) return p->row_sizeUV;
+                else            return 0;
+    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
+                if (p->pitchUV) {
+                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
+                        if (r < p->pitchUV)
+                                return r;
+                        return p->row_sizeUV;
+                } else return 0;
+    case AVS_PLANAR_Y_ALIGNED:
+                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
+                if (r <= p->pitch)
+                        return r;
+                return p->row_size;
+    }
+    return p->row_size;
+}
+
+AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
+        return p->height;}
+
+AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
+        switch (plane) {
+                case AVS_PLANAR_U: case AVS_PLANAR_V:
+                        if (p->pitchUV) return p->heightUV;
+                        return 0;
+        }
+        return p->height;}
+
+AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
+        return p->vfb->data + p->offset;}
+
+AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        switch (plane) {
+                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                default:           return p->vfb->data + p->offset;}
+}
+
+AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
+        return (p->refcount == 1 && p->vfb->refcount == 1);}
+
+AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
+{
+        if (avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else
+                return 0;
+}
+
+AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else if (plane==AVS_PLANAR_Y) {
+                return 0;
+        } else {
+                switch (plane) {
+                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                        default:       return p->vfb->data + p->offset;
+                }
+        }
+}
+
+
+AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
+// makes a shallow copy of a video frame
+AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
+  {avs_release_video_frame(f);}
+AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
+  {return avs_copy_video_frame(f);}
+#endif
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Value
+//
+
+// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
+// and avs_release_value appropiaty as you would if AVS_Value was
+// a pointer.
+
+// To maintain source code compatibility with future versions of the
+// avisynth_c API don't use the AVS_Value directly.  Use the helper
+// functions below.
+
+// AVS_Value is layed out identicly to AVSValue
+typedef struct AVS_Value AVS_Value;
+struct AVS_Value {
+  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
+               // for some function e'rror
+  short array_size;
+  union {
+    void * clip; // do not use directly, use avs_take_clip
+    char boolean;
+    int integer;
+    float floating_pt;
+    const char * string;
+    const AVS_Value * array;
+  } d;
+};
+
+// AVS_Value should be initilized with avs_void.
+// Should also set to avs_void after the value is released
+// with avs_copy_value.  Consider it the equalvent of setting
+// a pointer to NULL
+static const AVS_Value avs_void = {'v'};
+
+AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
+AVSC_API(void, avs_release_value)(AVS_Value);
+
+AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
+AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
+AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
+AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
+AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
+AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
+AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
+AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
+
+AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
+AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
+
+AVSC_INLINE int avs_as_bool(AVS_Value v)
+        { return v.d.boolean; }
+AVSC_INLINE int avs_as_int(AVS_Value v)
+        { return v.d.integer; }
+AVSC_INLINE const char * avs_as_string(AVS_Value v)
+        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
+AVSC_INLINE double avs_as_float(AVS_Value v)
+        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
+AVSC_INLINE const char * avs_as_error(AVS_Value v)
+        { return avs_is_error(v) ? v.d.string : 0; }
+AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
+        { return v.d.array; }
+AVSC_INLINE int avs_array_size(AVS_Value v)
+        { return avs_is_array(v) ? v.array_size : 1; }
+AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
+        { return avs_is_array(v) ? v.d.array[index] : v; }
+
+// only use these functions on an AVS_Value that does not already have
+// an active value.  Remember, treat AVS_Value as a fat pointer.
+AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
+        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+AVSC_INLINE AVS_Value avs_new_value_int(int v0)
+        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
+        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_float(float v0)
+        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
+AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
+        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
+        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
+#endif
+AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
+        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Clip
+//
+
+AVSC_API(void, avs_release_clip)(AVS_Clip *);
+AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
+
+AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
+
+AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
+
+AVSC_API(int, avs_get_version)(AVS_Clip *);
+
+AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
+// The returned video frame must be released with avs_release_video_frame
+
+AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
+// return field parity if field_based, else parity of first field in frame
+
+AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
+                             INT64 start, INT64 count);
+// start and count are in samples
+
+AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
+                                   int cachehints, int frame_range);
+
+// This is the callback type used by avs_add_function
+typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
+                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
+
+typedef struct AVS_FilterInfo AVS_FilterInfo;
+struct AVS_FilterInfo
+{
+  // these members should not be modified outside of the AVS_ApplyFunc callback
+  AVS_Clip * child;
+  AVS_VideoInfo vi;
+  AVS_ScriptEnvironment * env;
+  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
+                                  INT64 start, INT64 count);
+  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
+                                        int frame_range);
+  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
+
+  // Should be set when ever there is an error to report.
+  // It is cleared before any of the above methods are called
+  const char * error;
+  // this is to store whatever and may be modified at will
+  void * user_data;
+};
+
+// Create a new filter
+// fi is set to point to the AVS_FilterInfo so that you can
+//   modify it once it is initilized.
+// store_child should generally be set to true.  If it is not
+//    set than ALL methods (the function pointers) must be defined
+// If it is set than you do not need to worry about freeing the child
+//    clip.
+AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
+                                       AVS_FilterInfo * * fi,
+                                       AVS_Value child, int store_child);
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_ScriptEnvironment
+//
+
+// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
+enum {
+                                /* slowest CPU to support extension */
+  AVS_CPU_FORCE        = 0x01,   // N/A
+  AVS_CPU_FPU          = 0x02,   // 386/486DX
+  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
+  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
+  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
+  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
+  AVS_CPU_3DNOW        = 0x40,   // K6-2
+  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
+  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
+                                 // which only Hammer will have anyway)
+  AVS_CPUF_SSE3       = 0x100,   //  PIV+, K8 Venice
+  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
+  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
+  AVS_CPUF_SSE4_1     = 0x400,
+  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
+};
+
+AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
+
+AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
+
+AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
+AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
+
+AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
+ // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
+
+AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
+                                const char * name, const char * params,
+                                AVS_ApplyFunc apply, void * user_data);
+
+AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
+
+AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
+                               AVS_Value args, const char** arg_names);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
+
+AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
+
+//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
+//void avs_pop_context(AVS_ScriptEnvironment *);
+
+AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
+                                          const AVS_VideoInfo * vi, int align);
+// align should be at least 16
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE
+AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
+                                     const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+
+AVSC_INLINE
+AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
+                               const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+#endif
+
+
+AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
+
+AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
+
+typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
+AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
+
+AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
+// The returned video frame must be be released
+
+AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
+
+AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
+
+// avisynth.dll exports this; it's a way to use it as a library, without
+// writing an AVS script or without going through AVIFile.
+AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
+
+// this symbol is the entry point for the plugin and must
+// be defined
+AVSC_EXPORT
+const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
+
+
+AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
+
+
+AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
+// The returned video frame must be be released
+
+#ifdef AVSC_NO_DECLSPEC
+// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
+/*
+  The following functions needs to have been declared, probably from windows.h
+
+  void* malloc(size_t)
+  void free(void*);
+
+  HMODULE LoadLibrary(const char*);
+  void* GetProcAddress(HMODULE, const char*);
+  FreeLibrary(HMODULE);
+*/
+
+
+typedef struct AVS_Library AVS_Library;
+
+#define AVSC_DECLARE_FUNC(name) name##_func name
+
+struct AVS_Library {
+  HMODULE handle;
+
+  AVSC_DECLARE_FUNC(avs_add_function);
+  AVSC_DECLARE_FUNC(avs_at_exit);
+  AVSC_DECLARE_FUNC(avs_bit_blt);
+  AVSC_DECLARE_FUNC(avs_check_version);
+  AVSC_DECLARE_FUNC(avs_clip_get_error);
+  AVSC_DECLARE_FUNC(avs_copy_clip);
+  AVSC_DECLARE_FUNC(avs_copy_value);
+  AVSC_DECLARE_FUNC(avs_copy_video_frame);
+  AVSC_DECLARE_FUNC(avs_create_script_environment);
+  AVSC_DECLARE_FUNC(avs_delete_script_environment);
+  AVSC_DECLARE_FUNC(avs_function_exists);
+  AVSC_DECLARE_FUNC(avs_get_audio);
+  AVSC_DECLARE_FUNC(avs_get_cpu_flags);
+  AVSC_DECLARE_FUNC(avs_get_error);
+  AVSC_DECLARE_FUNC(avs_get_frame);
+  AVSC_DECLARE_FUNC(avs_get_parity);
+  AVSC_DECLARE_FUNC(avs_get_var);
+  AVSC_DECLARE_FUNC(avs_get_version);
+  AVSC_DECLARE_FUNC(avs_get_video_info);
+  AVSC_DECLARE_FUNC(avs_invoke);
+  AVSC_DECLARE_FUNC(avs_make_writable);
+  AVSC_DECLARE_FUNC(avs_new_c_filter);
+  AVSC_DECLARE_FUNC(avs_new_video_frame_a);
+  AVSC_DECLARE_FUNC(avs_release_clip);
+  AVSC_DECLARE_FUNC(avs_release_value);
+  AVSC_DECLARE_FUNC(avs_release_video_frame);
+  AVSC_DECLARE_FUNC(avs_save_string);
+  AVSC_DECLARE_FUNC(avs_set_cache_hints);
+  AVSC_DECLARE_FUNC(avs_set_global_var);
+  AVSC_DECLARE_FUNC(avs_set_memory_max);
+  AVSC_DECLARE_FUNC(avs_set_to_clip);
+  AVSC_DECLARE_FUNC(avs_set_var);
+  AVSC_DECLARE_FUNC(avs_set_working_dir);
+  AVSC_DECLARE_FUNC(avs_sprintf);
+  AVSC_DECLARE_FUNC(avs_subframe);
+  AVSC_DECLARE_FUNC(avs_subframe_planar);
+  AVSC_DECLARE_FUNC(avs_take_clip);
+  AVSC_DECLARE_FUNC(avs_vsprintf);
+};
+
+#undef AVSC_DECLARE_FUNC
+
+
+AVSC_INLINE AVS_Library * avs_load_library() {
+  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
+  if (library == NULL)
+    return NULL;
+  library->handle = LoadLibrary("avisynth");
+  if (library->handle == NULL)
+    goto fail;
+
+#define __AVSC_STRINGIFY(x) #x
+#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
+#define AVSC_LOAD_FUNC(name) {\
+  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
+  if (library->name == NULL)\
+    goto fail;\
+}
+
+  AVSC_LOAD_FUNC(avs_add_function);
+  AVSC_LOAD_FUNC(avs_at_exit);
+  AVSC_LOAD_FUNC(avs_bit_blt);
+  AVSC_LOAD_FUNC(avs_check_version);
+  AVSC_LOAD_FUNC(avs_clip_get_error);
+  AVSC_LOAD_FUNC(avs_copy_clip);
+  AVSC_LOAD_FUNC(avs_copy_value);
+  AVSC_LOAD_FUNC(avs_copy_video_frame);
+  AVSC_LOAD_FUNC(avs_create_script_environment);
+  AVSC_LOAD_FUNC(avs_delete_script_environment);
+  AVSC_LOAD_FUNC(avs_function_exists);
+  AVSC_LOAD_FUNC(avs_get_audio);
+  AVSC_LOAD_FUNC(avs_get_cpu_flags);
+  AVSC_LOAD_FUNC(avs_get_error);
+  AVSC_LOAD_FUNC(avs_get_frame);
+  AVSC_LOAD_FUNC(avs_get_parity);
+  AVSC_LOAD_FUNC(avs_get_var);
+  AVSC_LOAD_FUNC(avs_get_version);
+  AVSC_LOAD_FUNC(avs_get_video_info);
+  AVSC_LOAD_FUNC(avs_invoke);
+  AVSC_LOAD_FUNC(avs_make_writable);
+  AVSC_LOAD_FUNC(avs_new_c_filter);
+  AVSC_LOAD_FUNC(avs_new_video_frame_a);
+  AVSC_LOAD_FUNC(avs_release_clip);
+  AVSC_LOAD_FUNC(avs_release_value);
+  AVSC_LOAD_FUNC(avs_release_video_frame);
+  AVSC_LOAD_FUNC(avs_save_string);
+  AVSC_LOAD_FUNC(avs_set_cache_hints);
+  AVSC_LOAD_FUNC(avs_set_global_var);
+  AVSC_LOAD_FUNC(avs_set_memory_max);
+  AVSC_LOAD_FUNC(avs_set_to_clip);
+  AVSC_LOAD_FUNC(avs_set_var);
+  AVSC_LOAD_FUNC(avs_set_working_dir);
+  AVSC_LOAD_FUNC(avs_sprintf);
+  AVSC_LOAD_FUNC(avs_subframe);
+  AVSC_LOAD_FUNC(avs_subframe_planar);
+  AVSC_LOAD_FUNC(avs_take_clip);
+  AVSC_LOAD_FUNC(avs_vsprintf);
+
+#undef __AVSC_STRINGIFY
+#undef AVSC_STRINGIFY
+#undef AVSC_LOAD_FUNC
+
+  return library;
+
+fail:
+  free(library);
+  return NULL;
+}
+
+AVSC_INLINE void avs_free_library(AVS_Library *library) {
+  if (library == NULL)
+    return;
+  FreeLibrary(library->handle);
+  free(library);
+}
+#endif
+
+#endif
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@@ -0,0 +1,727 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef __AVXSYNTH_C__
+#define __AVXSYNTH_C__
+
+#include "windowsPorts/windows2linux.h"
+#include <stdarg.h>
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#define AVSC_USE_STDCALL 1
+
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
+#endif
+
+#define AVSC_INLINE static __inline
+
+#ifdef AVISYNTH_C_EXPORTS
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// Constants
+//
+
+#ifndef __AVXSYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 3 };
+#endif
+
+enum {AVS_SAMPLE_INT8  = 1<<0,
+      AVS_SAMPLE_INT16 = 1<<1,
+      AVS_SAMPLE_INT24 = 1<<2,
+      AVS_SAMPLE_INT32 = 1<<3,
+      AVS_SAMPLE_FLOAT = 1<<4};
+
+enum {AVS_PLANAR_Y=1<<0,
+      AVS_PLANAR_U=1<<1,
+      AVS_PLANAR_V=1<<2,
+      AVS_PLANAR_ALIGNED=1<<3,
+      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
+
+  // Colorspace properties.
+enum {AVS_CS_BGR = 1<<28,
+      AVS_CS_YUV = 1<<29,
+      AVS_CS_INTERLEAVED = 1<<30,
+      AVS_CS_PLANAR = 1<<31};
+
+  // Specific colorformats
+enum {
+  AVS_CS_UNKNOWN = 0,
+  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
+  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
+  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
+  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
+};
+
+enum {
+  AVS_IT_BFF = 1<<0,
+  AVS_IT_TFF = 1<<1,
+  AVS_IT_FIELDBASED = 1<<2};
+
+enum {
+  AVS_FILTER_TYPE=1,
+  AVS_FILTER_INPUT_COLORSPACE=2,
+  AVS_FILTER_OUTPUT_TYPE=9,
+  AVS_FILTER_NAME=4,
+  AVS_FILTER_AUTHOR=5,
+  AVS_FILTER_VERSION=6,
+  AVS_FILTER_ARGS=7,
+  AVS_FILTER_ARGS_INFO=8,
+  AVS_FILTER_ARGS_DESCRIPTION=10,
+  AVS_FILTER_DESCRIPTION=11};
+
+enum {  //SUBTYPES
+  AVS_FILTER_TYPE_AUDIO=1,
+  AVS_FILTER_TYPE_VIDEO=2,
+  AVS_FILTER_OUTPUT_TYPE_SAME=3,
+  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
+
+enum {
+  AVS_CACHE_NOTHING=0,
+  AVS_CACHE_RANGE=1,
+  AVS_CACHE_ALL=2,
+  AVS_CACHE_AUDIO=3,
+  AVS_CACHE_AUDIO_NONE=4,
+  AVS_CACHE_AUDIO_AUTO=5
+};
+
+#define AVS_FRAME_ALIGN 16
+
+typedef struct AVS_Clip AVS_Clip;
+typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoInfo
+//
+
+// AVS_VideoInfo is layed out identicly to VideoInfo
+typedef struct AVS_VideoInfo {
+  int width, height;    // width=0 means no video
+  unsigned fps_numerator, fps_denominator;
+  int num_frames;
+
+  int pixel_type;
+
+  int audio_samples_per_second;   // 0 means no audio
+  int sample_type;
+  INT64 num_audio_samples;
+  int nchannels;
+
+  // Imagetype properties
+
+  int image_type;
+} AVS_VideoInfo;
+
+// useful functions of the above
+AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
+        { return (p->width!=0); }
+
+AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
+        { return (p->audio_samples_per_second!=0); }
+
+AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_BGR); }
+
+AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
+        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
+
+AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
+
+AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_YUV ); }
+
+AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
+
+AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
+        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
+
+AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
+        { return ((p->pixel_type & c_space) == c_space); }
+
+AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
+        { return ((p->pixel_type & property)==property ); }
+
+AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type & AVS_CS_PLANAR); }
+
+AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_FIELDBASED); }
+
+AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
+        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
+
+AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_BFF); }
+
+AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_TFF); }
+
+AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
+{
+  switch (p->pixel_type) {
+      case AVS_CS_BGR24: return 24;
+      case AVS_CS_BGR32: return 32;
+      case AVS_CS_YUY2:  return 16;
+      case AVS_CS_YV12:
+      case AVS_CS_I420:  return 12;
+      default:           return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
+        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
+
+AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
+        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
+
+AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
+        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
+
+AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
+        { return p->audio_samples_per_second; }
+
+
+AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
+{
+    switch (p->sample_type) {
+      case AVS_SAMPLE_INT8:  return sizeof(signed char);
+      case AVS_SAMPLE_INT16: return sizeof(signed short);
+      case AVS_SAMPLE_INT24: return 3;
+      case AVS_SAMPLE_INT32: return sizeof(signed int);
+      case AVS_SAMPLE_FLOAT: return sizeof(float);
+      default: return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
+        { return p->nchannels*avs_bytes_per_channel_sample(p);}
+
+AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
+        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
+
+AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
+
+AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
+        { return bytes / avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return samples * avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
+        { return p->nchannels; }
+
+AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
+        { return p->sample_type;}
+
+// useful mutator
+AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
+        { p->image_type|=property; }
+
+AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
+        { p->image_type&=~property; }
+
+AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
+        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
+
+AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
+{
+    unsigned x=numerator, y=denominator;
+    while (y) {   // find gcd
+      unsigned t = x%y; x = y; y = t;
+    }
+    p->fps_numerator = numerator/x;
+    p->fps_denominator = denominator/x;
+}
+
+AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
+{
+        return (x->pixel_type == y->pixel_type)
+                || (avs_is_yv12(x) && avs_is_yv12(y));
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoFrame
+//
+
+// VideoFrameBuffer holds information about a memory block which is used
+// for video data.  For efficiency, instances of this class are not deleted
+// when the refcount reaches zero; instead they're stored in a linked list
+// to be reused.  The instances are deleted when the corresponding AVS
+// file is closed.
+
+// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrameBuffer {
+  unsigned char * data;
+  int data_size;
+  // sequence_number is incremented every time the buffer is changed, so
+  // that stale views can tell they're no longer valid.
+  long sequence_number;
+
+  long refcount;
+} AVS_VideoFrameBuffer;
+
+// VideoFrame holds a "window" into a VideoFrameBuffer.
+
+// AVS_VideoFrame is layed out identicly to IVideoFrame
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrame {
+  int refcount;
+  AVS_VideoFrameBuffer * vfb;
+  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
+} AVS_VideoFrame;
+
+// Access functions for AVS_VideoFrame
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return p->pitch;}
+
+AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
+  switch (plane) {
+  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
+  return p->pitch;}
+
+AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
+        return p->row_size; }
+
+AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
+        int r;
+    switch (plane) {
+    case AVS_PLANAR_U: case AVS_PLANAR_V:
+                if (p->pitchUV) return p->row_size>>1;
+                else            return 0;
+    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
+                if (p->pitchUV) {
+                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
+                        if (r < p->pitchUV)
+                                return r;
+                        return p->row_size>>1;
+                } else return 0;
+    case AVS_PLANAR_Y_ALIGNED:
+                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
+                if (r <= p->pitch)
+                        return r;
+                return p->row_size;
+    }
+    return p->row_size;
+}
+
+AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
+        return p->height;}
+
+AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
+        switch (plane) {
+                case AVS_PLANAR_U: case AVS_PLANAR_V:
+                        if (p->pitchUV) return p->height>>1;
+                        return 0;
+        }
+        return p->height;}
+
+AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
+        return p->vfb->data + p->offset;}
+
+AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        switch (plane) {
+                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                default:           return p->vfb->data + p->offset;}
+}
+
+AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
+        return (p->refcount == 1 && p->vfb->refcount == 1);}
+
+AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
+{
+        if (avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else
+                return 0;
+}
+
+AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else if (plane==AVS_PLANAR_Y) {
+                return 0;
+        } else {
+                switch (plane) {
+                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                        default:       return p->vfb->data + p->offset;
+                }
+        }
+}
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
+// makes a shallow copy of a video frame
+AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
+  {avs_release_video_frame(f);}
+AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
+  {return avs_copy_video_frame(f);}
+#endif
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Value
+//
+
+// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
+// and avs_release_value appropiaty as you would if AVS_Value was
+// a pointer.
+
+// To maintain source code compatibility with future versions of the
+// avisynth_c API don't use the AVS_Value directly.  Use the helper
+// functions below.
+
+// AVS_Value is layed out identicly to AVSValue
+typedef struct AVS_Value AVS_Value;
+struct AVS_Value {
+  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
+               // for some function e'rror
+  short array_size;
+  union {
+    void * clip; // do not use directly, use avs_take_clip
+    char boolean;
+    int integer;
+    INT64 integer64; // match addition of __int64 to avxplugin.h
+    float floating_pt;
+    const char * string;
+    const AVS_Value * array;
+  } d;
+};
+
+// AVS_Value should be initilized with avs_void.
+// Should also set to avs_void after the value is released
+// with avs_copy_value.  Consider it the equalvent of setting
+// a pointer to NULL
+static const AVS_Value avs_void = {'v'};
+
+AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
+AVSC_API(void, avs_release_value)(AVS_Value);
+
+AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
+AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
+AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
+AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
+AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
+AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
+AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
+AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
+AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+AVSC_INLINE int avs_as_bool(AVS_Value v)
+        { return v.d.boolean; }
+AVSC_INLINE int avs_as_int(AVS_Value v)
+        { return v.d.integer; }
+AVSC_INLINE const char * avs_as_string(AVS_Value v)
+        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
+AVSC_INLINE double avs_as_float(AVS_Value v)
+        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
+AVSC_INLINE const char * avs_as_error(AVS_Value v)
+        { return avs_is_error(v) ? v.d.string : 0; }
+AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
+        { return v.d.array; }
+AVSC_INLINE int avs_array_size(AVS_Value v)
+        { return avs_is_array(v) ? v.array_size : 1; }
+AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
+        { return avs_is_array(v) ? v.d.array[index] : v; }
+
+// only use these functions on am AVS_Value that does not already have
+// an active value.  Remember, treat AVS_Value as a fat pointer.
+AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
+        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+AVSC_INLINE AVS_Value avs_new_value_int(int v0)
+        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
+        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_float(float v0)
+        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
+AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
+        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
+        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
+#endif
+AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
+        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Clip
+//
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_clip)(AVS_Clip *);
+AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
+
+AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
+
+AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
+
+AVSC_API(int, avs_get_version)(AVS_Clip *);
+
+AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
+// The returned video frame must be released with avs_release_video_frame
+
+AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
+// return field parity if field_based, else parity of first field in frame
+
+AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
+                                  INT64 start, INT64 count);
+// start and count are in samples
+
+AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
+                                        int cachehints, size_t frame_range);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// This is the callback type used by avs_add_function
+typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
+                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
+
+typedef struct AVS_FilterInfo AVS_FilterInfo;
+struct AVS_FilterInfo
+{
+  // these members should not be modified outside of the AVS_ApplyFunc callback
+  AVS_Clip * child;
+  AVS_VideoInfo vi;
+  AVS_ScriptEnvironment * env;
+  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
+                                  INT64 start, INT64 count);
+  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
+                                        int frame_range);
+  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
+
+  // Should be set when ever there is an error to report.
+  // It is cleared before any of the above methods are called
+  const char * error;
+  // this is to store whatever and may be modified at will
+  void * user_data;
+};
+
+// Create a new filter
+// fi is set to point to the AVS_FilterInfo so that you can
+//   modify it once it is initilized.
+// store_child should generally be set to true.  If it is not
+//    set than ALL methods (the function pointers) must be defined
+// If it is set than you do not need to worry about freeing the child
+//    clip.
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
+                                      AVS_FilterInfo * * fi,
+                                      AVS_Value child, int store_child);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_ScriptEnvironment
+//
+
+// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
+enum {
+                                /* slowest CPU to support extension */
+  AVS_CPU_FORCE        = 0x01,   // N/A
+  AVS_CPU_FPU          = 0x02,   // 386/486DX
+  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
+  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
+  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
+  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
+  AVS_CPU_3DNOW        = 0x40,   // K6-2
+  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
+  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
+                                 // which only Hammer will have anyway)
+};
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
+
+AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
+
+AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
+AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
+
+AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
+ // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
+
+AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
+                                     const char * name, const char * params,
+                                     AVS_ApplyFunc apply, void * user_data);
+
+AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
+
+AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
+                               AVS_Value args, const char** arg_names);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
+
+AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
+
+//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
+//void avs_pop_context(AVS_ScriptEnvironment *);
+
+AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
+                                          const AVS_VideoInfo * vi, int align);
+// align should be at least 16
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE
+AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
+                                     const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+
+AVSC_INLINE
+AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
+                               const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+#endif
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
+
+AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
+
+typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
+AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
+
+AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
+// The returned video frame must be be released
+
+AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
+
+AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
+
+// avisynth.dll exports this; it's a way to use it as a library, without
+// writing an AVS script or without going through AVIFile.
+AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// this symbol is the entry point for the plugin and must
+// be defined
+AVSC_EXPORT
+const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
+
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
+
+
+AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
+// The returned video frame must be be released
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#endif //__AVXSYNTH_C__
--- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h
+++ b/compat/avisynth/windowsPorts/basicDataTypeConversions.h
@@ -0,0 +1,85 @@
+#ifndef __DATA_TYPE_CONVERSIONS_H__
+#define __DATA_TYPE_CONVERSIONS_H__
+
+#include <stdint.h>
+#include <wchar.h>
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+
+typedef int64_t __int64;
+typedef int32_t __int32;
+#ifdef __cplusplus
+typedef bool BOOL;
+#else
+typedef uint32_t BOOL;
+#endif // __cplusplus
+typedef void* HMODULE;
+typedef void* LPVOID;
+typedef void* PVOID;
+typedef PVOID HANDLE;
+typedef HANDLE HWND;
+typedef HANDLE HINSTANCE;
+typedef void* HDC;
+typedef void* HBITMAP;
+typedef void* HICON;
+typedef void* HFONT;
+typedef void* HGDIOBJ;
+typedef void* HBRUSH;
+typedef void* HMMIO;
+typedef void* HACMSTREAM;
+typedef void* HACMDRIVER;
+typedef void* HIC;
+typedef void* HACMOBJ;
+typedef HACMSTREAM* LPHACMSTREAM;
+typedef void* HACMDRIVERID;
+typedef void* LPHACMDRIVER;
+typedef unsigned char BYTE;
+typedef BYTE* LPBYTE;
+typedef char TCHAR;
+typedef TCHAR* LPTSTR;
+typedef const TCHAR* LPCTSTR;
+typedef char* LPSTR;
+typedef LPSTR LPOLESTR;
+typedef const char* LPCSTR;
+typedef LPCSTR LPCOLESTR;
+typedef wchar_t WCHAR;
+typedef unsigned short WORD;
+typedef unsigned int UINT;
+typedef UINT MMRESULT;
+typedef uint32_t DWORD;
+typedef DWORD COLORREF;
+typedef DWORD FOURCC;
+typedef DWORD HRESULT;
+typedef DWORD* LPDWORD;
+typedef DWORD* DWORD_PTR;
+typedef int32_t LONG;
+typedef int32_t* LONG_PTR;
+typedef LONG_PTR LRESULT;
+typedef uint32_t ULONG;
+typedef uint32_t* ULONG_PTR;
+//typedef __int64_t intptr_t;
+typedef uint64_t _fsize_t;
+
+
+//
+// Structures
+//
+
+typedef struct _GUID {
+  DWORD Data1;
+  WORD  Data2;
+  WORD  Data3;
+  BYTE  Data4[8];
+} GUID;
+
+typedef GUID REFIID;
+typedef GUID CLSID;
+typedef CLSID* LPCLSID;
+typedef GUID IID;
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+#endif //  __DATA_TYPE_CONVERSIONS_H__
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -0,0 +1,77 @@
+#ifndef __WINDOWS2LINUX_H__
+#define __WINDOWS2LINUX_H__
+
+/*
+ * LINUX SPECIFIC DEFINITIONS
+*/
+//
+// Data types conversions
+//
+#include <stdlib.h>
+#include <string.h>
+#include "basicDataTypeConversions.h"
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+//
+// purposefully define the following MSFT definitions
+// to mean nothing (as they do not mean anything on Linux)
+//
+#define __stdcall
+#define __cdecl
+#define noreturn
+#define __declspec(x)
+#define STDAPI       extern "C" HRESULT
+#define STDMETHODIMP HRESULT __stdcall
+#define STDMETHODIMP_(x) x __stdcall
+
+#define STDMETHOD(x)    virtual HRESULT x
+#define STDMETHOD_(a, x) virtual a x
+
+#ifndef TRUE
+#define TRUE  true
+#endif
+
+#ifndef FALSE
+#define FALSE false
+#endif
+
+#define S_OK                (0x00000000)
+#define S_FALSE             (0x00000001)
+#define E_NOINTERFACE       (0X80004002)
+#define E_POINTER           (0x80004003)
+#define E_FAIL              (0x80004005)
+#define E_OUTOFMEMORY       (0x8007000E)
+
+#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
+#define FAILED(hr)              ((hr) & 0x80000000)
+#define SUCCEEDED(hr)           (!FAILED(hr))
+
+
+//
+// Functions
+//
+#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
+#define MAKEWORD(a,b) ((a << 8) | (b))
+
+#define lstrlen                             strlen
+#define lstrcpy                             strcpy
+#define lstrcmpi                            strcasecmp
+#define _stricmp                            strcasecmp
+#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
+#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
+// Windows uses (new, old) ordering but GCC has (old, new)
+#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
+
+#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
+#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
+#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
+
+#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+
+#endif //  __WINDOWS2LINUX_H__
--- a/compat/getopt.c
+++ b/compat/getopt.c
@@ -46,7 +46,7 @@ static int getopt(int argc, char *argv[], char *opts)
    int c;
    char *cp;

-    if (sp == 1)
+    if (sp == 1) {
        if (optind >= argc ||
            argv[optind][0] != '-' || argv[optind][1] == '\0')
            return EOF;
@@ -54,6 +54,7 @@ static int getopt(int argc, char *argv[], char *opts)
            optind++;
            return EOF;
        }
+    }
    optopt = c = argv[optind][sp];
    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
        fprintf(stderr, ": illegal option -- %c\n", c);
--- a/libavcodec/os2threads.h
+++ b/libavcodec/os2threads.h
--- a/compat/plan9/head
+++ b/compat/plan9/head
@@ -0,0 +1,10 @@
+#!/bin/sh
+
+n=10
+
+case "$1" in
+    -n)  n=$2;      shift 2 ;;
+    -n*) n=${1#-n}; shift   ;;
+esac
+
+exec sed ${n}q "$@"
--- a/compat/plan9/main.c
+++ b/compat/plan9/main.c
@@ -0,0 +1,34 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+int plan9_main(int argc, char **argv);
+
+#undef main
+int main(int argc, char **argv)
+{
+    /* The setfcr() function in lib9 is broken, must use asm. */
+#ifdef __i386
+    short fcr;
+    __asm__ volatile ("fstcw        %0 \n"
+                      "or      $63, %0 \n"
+                      "fldcw        %0 \n"
+                      : "=m"(fcr));
+#endif
+
+    return plan9_main(argc, argv);
+}
--- a/compat/plan9/printf
+++ b/compat/plan9/printf
@@ -0,0 +1,2 @@
+#!/bin/sh
+exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"
--- a/compat/strtod.c
+++ b/compat/strtod.c
@@ -19,7 +19,6 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

-#include <ctype.h>
 #include <limits.h>
 #include <stdlib.h>

@@ -49,7 +48,7 @@ double avpriv_strtod(const char *nptr, char **endptr)
    double res;

    /* Skip leading spaces */
-    while (isspace(*nptr))
+    while (av_isspace(*nptr))
        nptr++;

    if (!av_strncasecmp(nptr, "infinity", 8)) {
--- a/compat/tms470/math.h
+++ b/compat/tms470/math.h
@@ -0,0 +1,7 @@
+#include_next <math.h>
+
+#undef INFINITY
+#undef NAN
+
+#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
+#define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
--- a/libavcodec/w32pthreads.h
+++ b/libavcodec/w32pthreads.h
@@ -4,20 +4,20 @@
 * Authors: Steven Walters <kemuri9@gmail.com>
 *          Pegasys Inc. <http://www.pegasys-inc.com>
 *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
 *
- * Libav is free software; you can redistribute it and/or
+ * 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.
 *
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

@@ -26,8 +26,8 @@
 * w32threads to pthreads wrapper
 */

-#ifndef AVCODEC_W32PTHREADS_H
-#define AVCODEC_W32PTHREADS_H
+#ifndef LIBAV_W32PTHREADS_H
+#define LIBAV_W32PTHREADS_H

 /* Build up a pthread-like API using underlying Windows API. Have only static
 * methods so as to not conflict with a potentially linked in pthread-win32
@@ -43,7 +43,7 @@
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"

-typedef struct {
+typedef struct pthread_t {
    void *handle;
    void *(*func)(void* arg);
    void *arg;
@@ -57,7 +57,7 @@ typedef CRITICAL_SECTION pthread_mutex_t;
 /* This is the CONDITIONAL_VARIABLE typedef for using Window's native
 * conditional variables on kernels 6.0+.
 * MinGW does not currently have this typedef. */
-typedef struct {
+typedef struct pthread_cond_t {
    void *ptr;
 } pthread_cond_t;

@@ -118,7 +118,7 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)

 /* for pre-Windows 6.0 platforms we need to define and use our own condition
 * variable and api */
-typedef struct {
+typedef struct  win32_cond_t {
    pthread_mutex_t mtx_broadcast;
    pthread_mutex_t mtx_waiter_count;
    volatile int waiter_count;
@@ -257,6 +257,7 @@ static void pthread_cond_signal(pthread_cond_t *cond)

 static void w32thread_init(void)
 {
+#if _WIN32_WINNT < 0x0600
    HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
    /* if one is available, then they should all be available */
    cond_init      =
@@ -267,6 +268,13 @@ static void w32thread_init(void)
        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
    cond_wait      =
        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
+#else
+    cond_init      = InitializeConditionVariable;
+    cond_broadcast = WakeAllConditionVariable;
+    cond_signal    = WakeConditionVariable;
+    cond_wait      = SleepConditionVariableCS;
+#endif
+
 }

-#endif /* AVCODEC_W32PTHREADS_H */
+#endif /* LIBAV_W32PTHREADS_H */
--- a/2207
+++ b/2207
--- a/doc/APIchanges
+++ b/doc/APIchanges
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -44,7 +44,7 @@ PROJECT_LOGO           =
 # If a relative path is entered, it will be relative to the location
 # where doxygen was started. If left blank the current directory will be used.

-OUTPUT_DIRECTORY       = doxy
+OUTPUT_DIRECTORY       = doc/doxy

 # If the CREATE_SUBDIRS tag is set to YES, then doxygen will create
 # 4096 sub-directories (in 2 levels) under the output directory of each output
@@ -277,7 +277,7 @@ SUBGROUPING            = YES
 # be useful for C code in case the coding convention dictates that all compound
 # types are typedef'ed and only the typedef is referenced, never the tag name.

-TYPEDEF_HIDES_STRUCT   = NO
+TYPEDEF_HIDES_STRUCT   = YES

 # The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
 # determine which symbols to keep in memory and which to flush to disk.
@@ -288,7 +288,7 @@ TYPEDEF_HIDES_STRUCT   = NO
 # causing a significant performance penality.
 # If the system has enough physical memory increasing the cache will improve the
 # performance by keeping more symbols in memory. Note that the value works on
-# a logarithmic scale so increasing the size by one will rougly double the
+# a logarithmic scale so increasing the size by one will roughly double the
 # memory usage. The cache size is given by this formula:
 # 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
 # corresponding to a cache size of 2^16 = 65536 symbols
@@ -409,7 +409,7 @@ INLINE_INFO            = YES
 # alphabetically by member name. If set to NO the members will appear in
 # declaration order.

-SORT_MEMBER_DOCS       = YES
+SORT_MEMBER_DOCS       = NO

 # If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the
 # brief documentation of file, namespace and class members alphabetically
@@ -489,12 +489,6 @@ MAX_INITIALIZER_LINES  = 30

 SHOW_USED_FILES        = YES

-# If the sources in your project are distributed over multiple directories
-# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy
-# in the documentation. The default is NO.
-
-SHOW_DIRECTORIES       = NO
-
 # Set the SHOW_FILES tag to NO to disable the generation of the Files page.
 # This will remove the Files entry from the Quick Index and from the
 # Folder Tree View (if specified). The default is YES.
@@ -645,15 +639,14 @@ EXCLUDE_SYMBOLS        =
 # directories that contain example code fragments that are included (see
 # the \include command).

-EXAMPLE_PATH           = libavcodec/ \
-                         libavformat/
+EXAMPLE_PATH           = doc/examples/

 # If the value of the EXAMPLE_PATH tag contains directories, you can use the
 # EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
 # and *.h) to filter out the source-files in the directories. If left
 # blank all files are included.

-EXAMPLE_PATTERNS       = *-example.c
+EXAMPLE_PATTERNS       = *.c

 # If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be
 # searched for input files to be used with the \include or \dontinclude
@@ -716,7 +709,7 @@ INLINE_SOURCES         = NO
 # doxygen to hide any special comment blocks from generated source code
 # fragments. Normal C and C++ comments will always remain visible.

-STRIP_CODE_COMMENTS    = YES
+STRIP_CODE_COMMENTS    = NO

 # If the REFERENCED_BY_RELATION tag is set to YES
 # then for each documented function all documented
@@ -800,13 +793,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.

-HTML_HEADER            = doc/doxy/header.html
+#HTML_HEADER            = doc/doxy/header.html

 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.

-HTML_FOOTER            = doc/doxy/footer.html
+#HTML_FOOTER            = doc/doxy/footer.html

 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -815,7 +808,7 @@ HTML_FOOTER            = doc/doxy/footer.html
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!

-HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
+#HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css

 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
@@ -825,7 +818,7 @@ HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
 # 180 is cyan, 240 is blue, 300 purple, and 360 is red again.
 # The allowed range is 0 to 359.

-HTML_COLORSTYLE_HUE    = 120
+#HTML_COLORSTYLE_HUE    = 120

 # The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of
 # the colors in the HTML output. For a value of 0 the output will use
@@ -848,12 +841,6 @@ HTML_COLORSTYLE_GAMMA  = 80

 HTML_TIMESTAMP         = YES

-# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes,
-# files or namespaces will be aligned in HTML using tables. If set to
-# NO a bullet list will be used.
-
-HTML_ALIGN_MEMBERS     = YES
-
 # If the HTML_DYNAMIC_SECTIONS tag is set to YES then the generated HTML
 # documentation will contain sections that can be hidden and shown after the
 # page has loaded. For this to work a browser that supports
@@ -1034,11 +1021,6 @@ ENUM_VALUES_PER_LINE   = 4

 GENERATE_TREEVIEW      = NO

-# By enabling USE_INLINE_TREES, doxygen will generate the Groups, Directories,
-# and Class Hierarchy pages using a tree view instead of an ordered list.
-
-USE_INLINE_TREES       = NO
-
 # If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be
 # used to set the initial width (in pixels) of the frame in which the tree
 # is shown.
@@ -1374,14 +1356,9 @@ INCLUDE_FILE_PATTERNS  =
 # instead of the = operator.

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

 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
 # this tag can be used to specify a list of macro names that should be expanded.
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -1,13 +1,31 @@
-MANPAGES    = $(PROGS-yes:%=doc/%.1)
-PODPAGES    = $(PROGS-yes:%=doc/%.pod)
-HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
+LIBRARIES-$(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-$(CONFIG_AVUTIL)     += ffmpeg-utils
+COMPONENTS-$(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
+
+MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
+MANPAGES    = $(MANPAGES1) $(MANPAGES3)
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)  $(PROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(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                                         \
-              doc/syntax.html                                           \

 TXTPAGES    = doc/fate.txt                                              \

@@ -22,6 +40,7 @@ all-$(CONFIG_DOC): doc

 doc: documentation

+apidoc: doc/doxy/html
 documentation: $(DOCS)

 TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
@@ -41,18 +60,33 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 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 $@ $<
+	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+
+doc/%-all.html: TAG = HTML
+doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
+	$(Q)$(TEXIDEP)
+	$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<

 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(GENTEXI)
+doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
+	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@

-doc/%.1: TAG = MAN
+doc/%-all.pod: TAG = POD
+doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
+	$(Q)$(TEXIDEP)
+	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -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=" " $< > $@

-$(DOCS): | doc/
+$(DOCS) doc/doxy/html: | doc/
+
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^

 install-man:

@@ -61,17 +95,26 @@ install-progs-$(CONFIG_DOC): install-man

 install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
-	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
+	$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
+	$(Q)mkdir -p "$(MANDIR)/man3"
+	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
 endif

 uninstall: uninstall-man

 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(MANPAGES1))
+	$(RM) $(addprefix "$(MANDIR)/man3/",$(MANPAGES3))

-clean::
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
+clean:: docclean
+
+distclean:: docclean
+	$(RM) doc/config.texi
+
+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: doc documentation
+.PHONY: apidoc doc documentation
--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -1,7 +1,7 @@
 Release Notes
 =============

-* 0.10 "Freedom"  January, 2012
+* 2.0 "Nameless"  July, 2013


 General notes
@@ -14,3 +14,7 @@ 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.
+
+AVI/AVXSynth
+--------
+If you want to use FFmpeg with AVISynth, you need AVISynth 2.6.0 at minimum.
--- a/doc/authors.texi
+++ b/doc/authors.texi
@@ -0,0 +1,11 @@
+@chapter Authors
+
+The FFmpeg developers.
+
+For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+@command{git log} in the FFmpeg source directory, or browsing the
+online repository at @url{http://source.ffmpeg.org}.
+
+Maintainers for the specific components are listed in the file
+@file{MAINTAINERS} in the source code tree.
--- a/doc/avtools-common-opts.texi
+++ b/doc/avtools-common-opts.texi
@@ -1,31 +1,33 @@
-All the numerical options, if not specified otherwise, accept in input
-a string representing a number, which may contain one of the
-International System number postfixes, for example 'K', 'M', 'G'.
-If 'i' is appended after the postfix, powers of 2 are used instead of
-powers of 10. The 'B' postfix multiplies the value for 8, and can be
-appended after another postfix or used alone. This allows using for
-example 'KB', 'MiB', 'G' and 'B' as postfix.
+All the numerical options, if not specified otherwise, accept a string
+representing a number as input, which may be followed by one of the SI
+unit prefixes, for example: 'K', 'M', or 'G'.
+
+If 'i' is appended to the SI unit prefix, the complete prefix will be
+interpreted as a unit prefix for binary multiplies, which are based on
+powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
+prefix multiplies the value by 8. This allows using, for example:
+'KB', 'MiB', 'G' and 'B' as number suffixes.

 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".
+the option name with "no". For example using "-nofoo"
+will set the boolean option with name "foo" to false.

@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.
+are used to precisely specify which stream(s) a given option belongs 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
+separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
+@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
+A stream specifier can match several streams, so that the option is 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}
+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:
@@ -34,20 +36,21 @@ Possible forms of stream specifiers are:
 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
+@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
+'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
+stream number @var{stream_index} of this type. Otherwise, it 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.
+If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
+in the program with the id @var{program_id}. Otherwise, it matches all streams in the
+program.
@item #@var{stream_id}
-Matches the stream by format-specific ID.
+Matches the stream by a format-specific ID.
@end table

@section Generic options

-These options are shared amongst the av* tools.
+These options are shared amongst the ff* tools.

@table @option

@@ -76,6 +79,10 @@ Print detailed information about the demuxer named @var{demuxer_name}. Use the
 Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.

+@item filter=@var{filter_name}
+Print detailed information about the filter name @var{filter_name}. Use the
+@option{-filters} option to get a list of all filters.
+
@end table

@item -version
@@ -84,14 +91,6 @@ 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.

@@ -122,18 +121,36 @@ Show available sample formats.
@item -layouts
 Show channel names and standard channel layouts.

-@item -loglevel @var{loglevel} | -v @var{loglevel}
+@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
 Set the logging level used by the library.
+Adding "repeat+" indicates that repeated log output should not be compressed
+to the first line and the "Last message repeated n times" line will be
+omitted. "repeat" can also be used alone.
+If "repeat" is used alone, and with no prior loglevel set, the default
+loglevel will be used. If multiple loglevel parameters are given, using
+'repeat' will not change the loglevel.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
+Show nothing at all; be silent.
@item panic
+Only show fatal errors which could lead the process to crash, such as
+and assert failure. This is not currently used for anything.
@item fatal
+Only show fatal errors. These are errors after which the process absolutely
+cannot continue after.
@item error
+Show all errors, including ones which can be recovered from.
@item warning
+Show all warnings and errors. Any message related to possibly
+incorrect or unexpected events will be shown.
@item info
+Show informative messages during processing. This is in addition to
+warnings and errors. This is the default value.
@item verbose
+Same as @code{info}, except more verbose.
@item debug
+Show everything, including debugging information.
@end table

 By default the program logs to stderr, if coloring is supported by the
@@ -151,8 +168,21 @@ directory.
 This file can be useful for bug reports.
 It also implies @code{-loglevel verbose}.

-Note: setting the environment variable @code{FFREPORT} to any value has the
-same effect.
+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
@@ -162,7 +192,61 @@ ffmpeg -cpuflags -sse+mmx ...
 ffmpeg -cpuflags mmx ...
 ffmpeg -cpuflags 0 ...
@end example
+Possible flags for this option are:
+@table @samp
+@item x86
+@table @samp
+@item mmx
+@item mmxext
+@item sse
+@item sse2
+@item sse2slow
+@item sse3
+@item sse3slow
+@item ssse3
+@item atom
+@item sse4.1
+@item sse4.2
+@item avx
+@item xop
+@item fma4
+@item 3dnow
+@item 3dnowext
+@item cmov
+@end table
+@item ARM
+@table @samp
+@item armv5te
+@item armv6
+@item armv6t2
+@item vfp
+@item vfpv3
+@item neon
+@end table
+@item PowerPC
+@table @samp
+@item altivec
+@end table
+@item Specific Processors
+@table @samp
+@item pentium2
+@item pentium3
+@item pentium4
+@item k6
+@item k62
+@item athlon
+@item athlonxp
+@item k8
+@end table
+@end table

+@item -opencl_options options (@emph{global})
+Set OpenCL environment options. This option is only available when
+FFmpeg has been compiled with @code{--enable-opencl}.
+
+@var{options} must be a list of @var{key}=@var{value} option pairs
+separated by ':'. See the ``OpenCL Options'' section in the
+ffmpeg-utils manual for the list of supported options.
@end table

@section AVOptions
@@ -195,6 +279,3 @@ 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.
-
-@include avoptions_codec.texi
-@include avoptions_format.texi
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -17,8 +17,19 @@ Below is a description of the currently available bitstream filters.

@section aac_adtstoasc

+Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
+bitstream filter.
+
+This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
+ADTS header and removes the ADTS header.
+
+This is required for example when copying an AAC stream from a raw
+ADTS AAC container to a FLV or a MOV/MP4 file.
+
@section chomp

+Remove zero padding at the end of a packet.
+
@section dump_extradata

@section h264_mp4toannexb
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -60,4 +60,102 @@ This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
 documented.

+@section libcelt
+
+libcelt decoder wrapper.
+
+libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
+Requires the presence of the libcelt headers and library during configuration.
+You need to explicitly configure the build with @code{--enable-libcelt}.
+
+@section libgsm
+
+libgsm decoder wrapper.
+
+libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
+the presence of the libgsm headers and library during configuration. You need
+to explicitly configure the build with @code{--enable-libgsm}.
+
+This decoder supports both the ordinary GSM and the Microsoft variant.
+
+@section libilbc
+
+libilbc decoder wrapper.
+
+libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
+audio codec. Requires the presence of the libilbc headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libilbc}.
+
+@subsection Options
+
+The following option is supported by the libilbc wrapper.
+
+@table @option
+@item enhance
+
+Enable the enhancement of the decoded audio when set to 1. The default
+value is 0 (disabled).
+
+@end table
+
+@section libopencore-amrnb
+
+libopencore-amrnb decoder wrapper.
+
+libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
+Narrowband audio codec. Using it requires the presence of the
+libopencore-amrnb headers and library during configuration. You need to
+explicitly configure the build with @code{--enable-libopencore-amrnb}.
+
+An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
+without this library.
+
+@section libopencore-amrwb
+
+libopencore-amrwb decoder wrapper.
+
+libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
+Wideband audio codec. Using it requires the presence of the
+libopencore-amrwb headers and library during configuration. You need to
+explicitly configure the build with @code{--enable-libopencore-amrwb}.
+
+An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
+without this library.
+
+@section libopus
+
+libopus decoder wrapper.
+
+libopus allows libavcodec to decode the Opus Interactive Audio Codec.
+Requires the presence of the libopus headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopus}.
+
@c man end AUDIO DECODERS
+
+@chapter Subtitles Decoders
+@c man begin SUBTILES DECODERS
+
+@section dvdsub
+
+This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
+also be found in VobSub file pairs and in some Matroska files.
+
+@subsection Options
+
+@table @option
+@item palette
+Specify the global palette used by the bitmaps. When stored in VobSub, the
+palette is normally specified in the index file; in Matroska, the palette is
+stored in the codec extra-data in the same format as in VobSub. In DVDs, the
+palette is stored in the IFO file, and therefore not available when reading
+from dumped VOB files.
+
+The format for this option is a string containing 16 24-bits hexadecimal
+numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
+ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
+7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
+@end table
+
+@c man end SUBTILES DECODERS
--- a/doc/default.css
+++ b/doc/default.css
@@ -0,0 +1,149 @@
+a {
+    color: #2D6198;
+}
+
+a:visited {
+    color: #884488;
+}
+
+#banner {
+    background-color: white;
+    position: relative;
+    text-align: center;
+}
+
+#banner img {
+    padding-bottom: 1px;
+    padding-top: 5px;
+}
+
+#body {
+    margin-left: 1em;
+    margin-right: 1em;
+}
+
+body {
+    background-color: #313131;
+    margin: 0;
+    text-align: justify;
+}
+
+.center {
+    margin-left: auto;
+    margin-right: auto;
+    text-align: center;
+}
+
+#container {
+    background-color: white;
+    color: #202020;
+    margin-left: 1em;
+    margin-right: 1em;
+}
+
+#footer {
+    text-align: center;
+}
+
+h1, h2, h3 {
+    padding-left: 0.4em;
+    border-radius: 4px;
+    padding-bottom: 0.2em;
+    padding-top: 0.2em;
+    border: 1px solid #6A996A;
+}
+
+h1 {
+    background-color: #7BB37B;
+    color: #151515;
+    font-size: 1.2em;
+    padding-bottom: 0.3em;
+    padding-top: 0.3em;
+}
+
+h2 {
+    color: #313131;
+    font-size: 0.9em;
+    background-color: #ABE3AB;
+}
+
+h3 {
+    color: #313131;
+    font-size: 0.8em;
+    margin-bottom: -8px;
+    background-color: #BBF3BB;
+}
+
+img {
+    border: 0;
+}
+
+#navbar {
+    background-color: #738073;
+    border-bottom: 1px solid #5C665C;
+    border-top: 1px solid #5C665C;
+    margin-top: 12px;
+    padding: 0.3em;
+    position: relative;
+    text-align: center;
+}
+
+#navbar a, #navbar_secondary a {
+    color: white;
+    padding: 0.3em;
+    text-decoration: none;
+}
+
+#navbar a:hover, #navbar_secondary a:hover {
+    background-color: #313131;
+    color: white;
+    text-decoration: none;
+}
+
+#navbar_secondary {
+    background-color: #738073;
+    border-bottom: 1px solid #5C665C;
+    border-left: 1px solid #5C665C;
+    border-right: 1px solid #5C665C;
+    padding: 0.3em;
+    position: relative;
+    text-align: center;
+}
+
+p {
+    margin-left: 1em;
+    margin-right: 1em;
+}
+
+pre {
+    margin-left: 3em;
+    margin-right: 3em;
+    padding: 0.3em;
+    border: 1px solid #bbb;
+    background-color: #f7f7f7;
+}
+
+dl dt {
+    font-weight: bold;
+}
+
+#proj_desc {
+    font-size: 1.2em;
+}
+
+#repos {
+    margin-left: 1em;
+    margin-right: 1em;
+    border-collapse: collapse;
+    border: solid 1px #6A996A;
+}
+
+#repos th {
+    background-color: #7BB37B;
+    border: solid 1px #6A996A;
+}
+
+#repos td {
+    padding: 0.2em;
+    border: solid 1px #6A996A;
+}
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -6,18 +6,128 @@ 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".
+configure option @code{--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}".
+@code{--disable-demuxers}, and selectively enable a single demuxer with
+the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
+with the option @code{--disable-demuxer=@var{DEMUXER}}.

-The option "-formats" of the ff* tools will display the list of
+The option @code{-formats} of the ff* tools will display the list of
 enabled demuxers.

 The description of some of the currently available demuxers follows.

+@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".
+
+@anchor{concat}
+@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.).
+
+The duration of each file is used to adjust the timestamps of the next file:
+if the duration is incorrect (because it was computed using the bit-rate or
+because the file is truncated, for example), it can cause artifacts. The
+@code{duration} directive can be used to override the duration stored in
+each file.
+
+@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.
+
+All subsequent directives apply to that file.
+
+@item @code{ffconcat version 1.0}
+Identify the script type and version. It also sets the @option{safe} option
+to 1 if it was to its default -1.
+
+To make FFmpeg recognize the format automatically, this directive must
+appears exactly as is (no extra space or byte-order-mark) on the very first
+line of the script.
+
+@item @code{duration @var{dur}}
+Duration of the file. This information can be specified from the file;
+specifying it here may be more efficient or help if the information from the
+file is not available or accurate.
+
+If the duration is set for all files, then it is possible to seek in the
+whole concatenated video.
+
+@end table
+
+@subsection Options
+
+This demuxer accepts the following option:
+
+@table @option
+
+@item safe
+If set to 1, reject unsafe file paths. A file path is considered safe if it
+does not contain a protocol specification and is relative and all components
+only contain characters from the portable character set (letters, digits,
+period, underscore and hyphen) and have no period at the beginning of a
+component.
+
+If set to 0, any file name is accepted.
+
+The default is -1, it is equivalent to 1 if the format was automatically
+probed and 0 otherwise.
+
+@end table
+
+@section libgme
+
+The Game Music Emu library is a collection of video game music file emulators.
+
+See @url{http://code.google.com/p/game-music-emu/} for more information.
+
+Some files have multiple tracks. The demuxer will pick the first track by
+default. The @option{track_index} option can be used to select a different
+track. Track indexes start at 0. The demuxer exports the number of tracks as
+@var{tracks} meta data entry.
+
+For very large files, the @option{max_size} option may have to be adjusted.
+
+@section libquvi
+
+Play media from Internet services using the quvi project.
+
+The demuxer accepts a @option{format} option to request a specific quality. It
+is by default set to @var{best}.
+
+See @url{http://quvi.sourceforge.net/} for more information.
+
+FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
+enabled.
+
@section image2

 Image file demuxer.
@@ -35,7 +145,7 @@ 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.
+Set the frame rate 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
@@ -113,6 +223,10 @@ to read from. Default value is 0.
 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 ts_from_file
+If set to 1, will set frame timestamp to modification time of image file. Note
+that monotonity of timestamps is not provided: images go in the same order as
+without this option. Default value is 0.
@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.
@@ -143,16 +257,34 @@ ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
@end example
@end itemize

-@section applehttp
+@section rawvideo

-Apple HTTP Live Streaming demuxer.
+Raw video 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".
+This demuxer allows to read raw video data. Since there is no header
+specifying the assumed video parameters, the user must specify them
+in order to be able to decode the data correctly.
+
+This demuxer accepts the following options:
+@table @option
+
+@item framerate
+Set input video frame rate. Default value is 25.
+
+@item pixel_format
+Set the input video pixel format. Default value is @code{yuv420p}.
+
+@item video_size
+Set the input video size. This value must be specified explicitly.
+@end table
+
+For example to read a rawvideo file @file{input.raw} with
+@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
+size of @code{320x240}, and a frame rate of 10 images per second, use
+the command:
+@example
+ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
+@end example

@section sbg

@@ -184,4 +316,25 @@ 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.

-@c man end INPUT DEVICES
+@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 DEMUXERS
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -147,30 +147,45 @@ 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
+All names should be composed with underscores (_), not CamelCase. For example,
+@samp{avfilter_get_video_buffer} is an acceptable 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:
+There are the 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.
+For file-scope variables and functions declared as @code{static}, no prefix
+is required.
@item
-For variables and functions used internally by the library, @code{ff_} prefix
-should be used.
-For example, @samp{ff_w64_demuxer}.
+For variables and functions visible outside of file scope, but only used
+internally by a library, an @code{ff_} prefix should be used,
+e.g. @samp{ff_w64_demuxer}.
@item
-For variables and functions used internally across multiple libraries, use
-@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
+For variables and functions visible outside of file scope, used internally
+across multiple libraries, use @code{avpriv_} as prefix, 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.
+Each library has its own prefix for public symbols, in addition to the
+commonly used @code{av_} (@code{avformat_} for libavformat,
+@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
+Check the existing code and choose names accordingly.
+Note that some symbols without these prefixes are also exported for
+retro-compatibility reasons. These exceptions are declared in the
+@code{lib<name>/lib<name>.v} files.
@end itemize

-@subsection Miscellanous conventions
+Furthermore, name space reserved for the system should not be invaded.
+Identifiers ending in @code{_t} are reserved by
+@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
+Also avoid names starting with @code{__} or @code{_} followed by an uppercase
+letter as they are reserved by the C standard. Names starting with @code{_}
+are reserved at the file level and may not be used for externally visible
+symbols. If in doubt, just avoid names starting with @code{_} altogether.
+
+@subsection Miscellaneous conventions
@itemize @bullet
@item
 fprintf and printf are forbidden in libavformat and libavcodec,
@@ -190,8 +205,8 @@ set shiftwidth=4
 set softtabstop=4
 set cindent
 set cinoptions=(0
-" allow tabs in Makefiles
-autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
+" 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/
@@ -217,8 +232,13 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:

@enumerate
@item
-   Contributions should be licensed under the LGPL 2.1, including an
-   "or any later version" clause, or the MIT license.  GPL 2 including
+   Contributions should be licensed under the
+   @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
+   including an "or any later version" clause, or, if you prefer
+   a gift-style license, the
+   @uref{http://www.isc.org/software/license/, ISC} or
+   @uref{http://mit-license.org/, MIT} license.
+   @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
   an "or any later version" clause is also acceptable, but LGPL is
   preferred.
@item
@@ -228,6 +248,13 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
   You can commit unfinished stuff (for testing etc), but it must be disabled
   (#ifdef etc) by default so it does not interfere with other developers'
   work.
+@item
+   The commit message should have a short first line in the form of
+   a @samp{topic: short description} as a header, separated by a newline
+   from the body consisting of an explanation of why the change is necessary.
+   If the commit fixes a known bug on the bug tracker, the commit message
+   should include its bug ID. Referring to the issue on the bug tracker does
+   not exempt you from writing an excerpt of the bug in the commit message.
@item
   You do not have to over-test things. If it works for you, and you think it
   should work for others, then commit. If your code has problems
@@ -317,7 +344,8 @@ 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).
+    change (e.g. encoder bug fix that matters for the decoder). The third
+    component always starts at 100 to distinguish FFmpeg from Libav.
@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
@@ -333,8 +361,6 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:

 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

@@ -357,11 +383,6 @@ 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.

-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}
-
 It also helps quite a bit if you tell us what the patch does (for example
 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
 and has no lrint()')
@@ -369,6 +390,13 @@ and has no lrint()')
 Also please if you send several patches, send each patch as a separate mail,
 do not attach several unrelated patches to the same mail.

+Patches should be posted to the
+@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
+mailing list. Use @code{git send-email} when possible since it will properly
+send patches without requiring extra care. If you cannot, then send patches
+as base64-encoded attachments, so your patch is not trashed during
+transmission.
+
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -397,7 +425,7 @@ send a reminder by email. Your patch should eventually be dealt with.
    When adding new codec IDs, also add an entry to the codec descriptor
    list in @file{libavcodec/codec_desc.c}.
@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?
@@ -429,7 +457,7 @@ send a reminder by email. Your patch should eventually be dealt with.
    Was the patch generated with git format-patch or send-email?
@item
    Did you sign off your patch? (git commit -s)
-    See @url{http://kerneltrap.org/files/Jeremy/DCO.txt} for the meaning
+    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.
@item
    Did you provide a clear git commit log message?
@@ -450,8 +478,10 @@ send a reminder by email. Your patch should eventually be dealt with.
    other security issues?
@item
    Did you test your decoder or demuxer against damaged data? If no, see
-    tools/trasher and the noise bitstream filter. Your decoder or demuxer
-    should not crash or end in a (near) infinite loop when fed damaged data.
+    tools/trasher, the noise bitstream filter, and
+    @uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
+    should not crash, end in a (near) infinite loop, or allocate ridiculous
+    amounts of memory when fed damaged data.
@item
    Does the patch not mix functional and cosmetic changes?
@item
@@ -495,6 +525,9 @@ send a reminder by email. Your patch should eventually be dealt with.
    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.
+@item
+    Test your code with valgrind and or Address Sanitizer to ensure it's free
+    of leaks, out of array accesses, etc.
@end enumerate

@section Patch review process
@@ -547,4 +580,129 @@ 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.


+@subsection Visualizing Test Coverage
+
+The FFmpeg build system allows visualizing the test coverage in an easy
+manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
+the following steps:
+
+@enumerate
+@item
+    Configure to compile with instrumentation enabled:
+    @code{configure --toolchain=gcov}.
+@item
+    Run your test case, either manually or via FATE. This can be either
+    the full FATE regression suite, or any arbitrary invocation of any
+    front-end tool provided by FFmpeg, in any combination.
+@item
+    Run @code{make lcov} to generate coverage data in HTML format.
+@item
+    View @code{lcov/index.html} in your preferred HTML viewer.
+@end enumerate
+
+You can use the command @code{make lcov-reset} to reset the coverage
+measurements. You will need to rerun @code{make lcov} after running a
+new test.
+
+@subsection Using Valgrind
+
+The configure script provides a shortcut for using valgrind to spot bugs
+related to memory handling. Just add the option
+@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
+to your configure line, and reasonable defaults will be set for running
+FATE under the supervision of either the @strong{memcheck} or the
+@strong{massif} tool of the valgrind suite.
+
+In case you need finer control over how valgrind is invoked, use the
+@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
+your configure line instead.
+
+@anchor{Release process}
+@section Release process
+
+FFmpeg maintains a set of @strong{release branches}, which are the
+recommended deliverable for system integrators and distributors (such as
+Linux distributions, etc.). At regular times, a @strong{release
+manager} prepares, tests and publishes tarballs on the
+@url{http://ffmpeg.org} website.
+
+There are two kinds of releases:
+
+@enumerate
+@item
+    @strong{Major releases} always include the latest and greatest
+    features and functionality.
+@item
+    @strong{Point releases} are cut from @strong{release} branches,
+    which are named @code{release/X}, with @code{X} being the release
+    version number.
+@end enumerate
+
+Note that we promise to our users that shared libraries from any FFmpeg
+release never break programs that have been @strong{compiled} against
+previous versions of @strong{the same release series} in any case!
+
+However, from time to time, we do make API changes that require adaptations
+in applications. Such changes are only allowed in (new) major releases and
+require further steps such as bumping library version numbers and/or
+adjustments to the symbol versioning file. Please discuss such changes
+on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
+
+@anchor{Criteria for Point Releases}
+@subsection Criteria for Point Releases
+
+Changes that match the following criteria are valid candidates for
+inclusion into a point release:
+
+@enumerate
+@item
+    Fixes a security issue, preferably identified by a @strong{CVE
+    number} issued by @url{http://cve.mitre.org/}.
+@item
+    Fixes a documented bug in @url{https://trac.ffmpeg.org}.
+@item
+    Improves the included documentation.
+@item
+    Retains both source code and binary compatibility with previous
+    point releases of the same release branch.
+@end enumerate
+
+The order for checking the rules is (1 OR 2 OR 3) AND 4.
+
+
+@subsection Release Checklist
+
+The release process involves the following steps:
+
+@enumerate
+@item
+    Ensure that the @file{RELEASE} file contains the version number for
+    the upcoming release.
+@item
+    Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
+@item
+    Announce the intent to do a release to the mailing list.
+@item
+    Make sure all relevant security fixes have been backported. See
+    @url{https://ffmpeg.org/security.html}.
+@item
+    Ensure that the FATE regression suite still passes in the release
+    branch on at least @strong{i386} and @strong{amd64}
+    (cf. @ref{Regression tests}).
+@item
+    Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
+    supplementing files that contain @code{gpg} signatures
+@item
+    Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
+    push an annotated tag in the form @code{nX}, with @code{X}
+    containing the version number.
+@item
+    Propose and send a patch to the @strong{ffmpeg-devel} mailing list
+    with a news entry for the website.
+@item
+    Publish the news entry.
+@item
+    Send announcement to the mailing list.
+@end enumerate
+
@bye
--- a/doc/devices.texi
+++ b/doc/devices.texi
@@ -0,0 +1,21 @@
+@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
--- a/doc/doxy-wrapper.sh
+++ b/doc/doxy-wrapper.sh
@@ -0,0 +1,14 @@
+#!/bin/sh
+
+SRC_PATH="${1}"
+DOXYFILE="${2}"
+
+shift 2
+
+doxygen - <<EOF
+@INCLUDE        = ${DOXYFILE}
+INPUT           = $@
+HTML_HEADER     = ${SRC_PATH}/doc/doxy/header.html
+HTML_FOOTER     = ${SRC_PATH}/doc/doxy/footer.html
+HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
+EOF
--- a/doc/doxy/doxy_stylesheet.css
+++ b/doc/doxy/doxy_stylesheet.css
--- a/doc/doxy/footer.html
+++ b/doc/doxy/footer.html
@@ -1,10 +1,9 @@
- </div>
-
- <div id="footer">
-   Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
- </div>

+ <footer class="footer pagination-right">
+   <span class="label label-info">
+    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
+   </span>
+ </footer>
 </div>
-
 </body>
 </html>
--- a/doc/doxy/header.html
+++ b/doc/doxy/header.html
@@ -1,14 +1,16 @@
-<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
-<html xmlns="http://www.w3.org/1999/xhtml">
+<!DOCTYPE html>
+<html>
 <head>
-<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
 <meta http-equiv="X-UA-Compatible" content="IE=9"/>
 <!--BEGIN PROJECT_NAME--><title>$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 id="container">
+<div class="container">

-<div id="body">
- <div>
+<!--Header replace -->
+<div class="menu">
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -412,6 +412,296 @@ Selected by Encoder (default)

@end table

+@section libmp3lame
+
+LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.
+
+Requires the presence of the libmp3lame headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libmp3lame}.
+
+@subsection Option Mapping
+
+The following options are supported by the libmp3lame wrapper,
+the LAME-equivalent options follow the FFmpeg ones.
+
+@multitable @columnfractions .2 .2
+@item FFmpeg            @tab LAME
+@item b                 @tab b
+Set bitrate expressed in bits/s, LAME @code{bitrate} is expressed in
+kilobits/s.
+@item q                 @tab V
+Set quality setting for VBR.
+@item compression_level @tab q
+Set algorithm quality. Valid arguments are integers in the 0-9 range.
+@item reservoir         @tab N.A.
+Enable use of bit reservoir. LAME has this enabled by default.
+@item joint_stereo      @tab -m j
+Enable the encoder to use (on a frame by frame basis) either L/R
+stereo or mid/side stereo.
+@end multitable
+
+@section libopencore-amrnb
+
+OpenCORE Adaptive Multi-Rate Narrowband encoder.
+
+Requires the presence of the libopencore-amrnb headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopencore-amrnb --enable-version3}.
+
+This is a mono-only encoder. Officially it only supports 8000Hz sample rate,
+but you can override it by setting @option{strict} to @samp{unofficial} or
+lower.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set bitrate in bits per second. Only the following bitrates are supported,
+otherwise libavcodec will round to the nearest valid bitrate.
+
+@table @option
+@item 4750
+@item 5150
+@item 5900
+@item 6700
+@item 7400
+@item 7950
+@item 10200
+@item 12200
+@end table
+
+@item dtx
+Allow discontinuous transmission (generate comfort noise) when set to 1. The
+default value is 0 (disabled).
+
+@end table
+
+@section libtwolame
+
+TwoLAME MP2 encoder wrapper.
+
+Requires the presence of the libtwolame headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libtwolame}.
+
+@subsection Options Mapping
+
+The following options are supported by the libtwolame wrapper. The
+TwoLAME-equivalent options follow the FFmpeg ones and are in
+parentheses.
+
+@table @option
+@item b
+(b) Set bitrate in bits/s. Note that FFmpeg @code{b} option is
+expressed in bits/s, twolame @code{b} in kilobits/s. The default
+value is 128k.
+
+@item q
+(V) Set quality for experimental VBR support. Maximum value range is
+from -50 to 50, useful range is from -10 to 10.
+
+@item mode
+(mode) Set MPEG mode. Possible values:
+
+@table @samp
+@item auto
+Choose mode automatically based on the input. This is the default.
+@item stereo
+Stereo
+@item joint_stereo
+Joint stereo
+@item dual_channel
+Dual channel
+@item mono
+Mono
+@end table
+
+@item psymodel
+(psyc-mode) Set psychoacoustic model to use in encoding. The argument
+must be an integer between -1 and 4, inclusive. The higher the value,
+the better the quality. The default value is 3.
+
+@item energy_levels
+(energy) Enable energy levels extensions when set to 1. The default
+value is 0 (disabled).
+
+@item error_protection
+(protect) Enable CRC error protection when set to 1. The default value
+is 0 (disabled).
+
+@item copyright
+(copyright) Set MPEG audio copyright flag when set to 1. The default
+value is 0 (disabled).
+
+@item original
+(original) Set MPEG audio original flag when set to 1. The default
+value is 0 (disabled).
+
+@end table
+
+@section libvo-aacenc
+
+VisualOn AAC encoder.
+
+Requires the presence of the libvo-aacenc headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libvo-aacenc --enable-version3}.
+
+@subsection Options
+
+The VisualOn AAC encoder only support encoding AAC-LC and up to 2
+channels. It is also CBR-only. It is considered to be worse than the
+native experimental FFmpeg AAC encoder.
+
+@table @option
+
+@item b
+Bitrate.
+
+@end table
+
+@section libvo-amrwbenc
+
+VisualOn Adaptive Multi-Rate Wideband encoder.
+
+Requires the presence of the libvo-amrwbenc headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libvo-amrwbenc --enable-version3}.
+
+This is a mono-only encoder. Officially it only supports 16000Hz sample
+rate, but you can override it by setting @option{strict} to
+@samp{unofficial} or lower.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set bitrate in bits/s. Only the following bitrates are supported, otherwise
+libavcodec will round to the nearest valid bitrate.
+
+@table @samp
+@item 6600
+@item 8850
+@item 12650
+@item 14250
+@item 15850
+@item 18250
+@item 19850
+@item 23050
+@item 23850
+@end table
+
+@item dtx
+Allow discontinuous transmission (generate comfort noise) when set to 1. The
+default value is 0 (disabled).
+
+@end table
+
+@section libopus
+
+libopus Opus Interactive Audio Codec encoder wrapper.
+
+Requires the presence of the libopus headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopus}.
+
+@subsection Option Mapping
+
+Most libopus options are modeled after the @command{opusenc} utility from
+opus-tools. The following is an option mapping chart describing options
+supported by the libopus wrapper, and their @command{opusenc}-equivalent
+in parentheses.
+
+@table @option
+
+@item b (@emph{bitrate})
+Set the bit rate in bits/s.  FFmpeg's @option{b} option is
+expressed in bits/s, while @command{opusenc}'s @option{bitrate} in
+kilobits/s.
+
+@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr})
+Set VBR mode. The FFmpeg @option{vbr} option has the following
+valid arguments, with the their @command{opusenc} equivalent options
+in parentheses:
+
+@table @samp
+@item off (@emph{hard-cbr})
+Use constant bit rate encoding.
+
+@item on (@emph{vbr})
+Use variable bit rate encoding (the default).
+
+@item constrained (@emph{cvbr})
+Use constrained variable bit rate encoding.
+@end table
+
+@item compression_level (@emph{comp})
+Set encoding algorithm complexity. Valid options are integers in
+the 0-10 range. 0 gives the fastest encodes but lower quality, while 10
+gives the highest quality but slowest encoding. The default is 10.
+
+@item frame_duration (@emph{framesize})
+Set maximum frame size, or duration of a frame in milliseconds. The
+argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
+frame sizes achieve lower latency but less quality at a given bitrate.
+Sizes greater than 20ms are only interesting at fairly low bitrates.
+The default of FFmpeg is 10ms, but is 20ms in @command{opusenc}.
+
+@item packet_loss (@emph{expect-loss})
+Set expected packet loss percentage. The default is 0.
+
+@item application (N.A.)
+Set intended application type. Valid options are listed below:
+
+@table @samp
+@item voip
+Favor improved speech intelligibility.
+@item audio
+Favor faithfulness to the input (the default).
+@item lowdelay
+Restrict to only the lowest delay modes.
+@end table
+
+@item cutoff (N.A.)
+Set cutoff bandwidth in Hz. The argument must be exactly one of the
+following: 4000, 6000, 8000, 12000, or 20000, corresponding to
+narrowband, mediumband, wideband, super wideband, and fullband
+respectively. The default is 0 (cutoff disabled).
+
+@end table
+
+@section libwavpack
+
+A wrapper providing WavPack encoding through libwavpack.
+
+Only lossless mode using 32-bit integer samples is supported currently.
+The @option{compression_level} option can be used to control speed vs.
+compression tradeoff, with the values mapped to libwavpack as follows:
+
+@table @option
+
+@item 0
+Fast mode - corresponding to the wavpack @option{-f} option.
+
+@item 1
+Normal (default) settings.
+
+@item 2
+High quality - corresponding to the wavpack @option{-h} option.
+
+@item 3
+Very high quality - corresponding to the wavpack @option{-hh} option.
+
+@item 4-8
+Same as 3, but with extra processing enabled - corresponding to the wavpack
+@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as
+@option{-x6}.
+
+@end table
+
@c man end AUDIO ENCODERS

@chapter Video Encoders
@@ -420,6 +710,46 @@ Selected by Encoder (default)
 A description of some of the currently available video encoders
 follows.

+@section libtheora
+
+Theora format supported through libtheora.
+
+Requires the presence of the libtheora headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libtheora}.
+
+@subsection Options
+
+The following global options are mapped to internal libtheora options
+which affect the quality and the bitrate of the encoded stream.
+
+@table @option
+@item b
+Set the video bitrate, only works if the @code{qscale} flag in
+@option{flags} is not enabled.
+
+@item flags
+Used to enable constant quality mode encoding through the
+@option{qscale} flag, and to enable the @code{pass1} and @code{pass2}
+modes.
+
+@item g
+Set the GOP size.
+
+@item global_quality
+Set the global quality in lambda units, only works if the
+@code{qscale} flag in @option{flags} is enabled. The value is clipped
+in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3
+to get a value in the native libtheora range [0-63]. A higher value
+corresponds to a higher quality.
+
+For example, to set maximum constant quality encoding with
+@command{ffmpeg}:
+@example
+ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg
+@end example
+@end table
+
@section libvpx

 VP8 format supported through libvpx.
@@ -541,47 +871,183 @@ For more information about libvpx see:

@section libx264

-H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 format supported through
-libx264.
+x264 H.264/MPEG-4 AVC encoder wrapper.

 Requires the presence of the libx264 headers and library during
 configuration. You need to explicitly configure the build with
@code{--enable-libx264}.

-@subsection Options
+x264 supports an impressive number of features, including 8x8 and 4x4 adaptive
+spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding,
+interlacing (MBAFF), lossless mode, psy optimizations for detail retention
+(adaptive quantization, psy-RD, psy-trellis).

+The FFmpeg wrapper provides a mapping for most of them using global options
+that match those of the encoders and provides private options for the unique
+encoder options. Additionally an expert override is provided to directly pass
+a list of key=value tuples as accepted by x264_param_parse.
+
+@subsection Option Mapping
+
+The following options are supported by the x264 wrapper, the x264-equivalent
+options follow the FFmpeg ones.
+
+@multitable @columnfractions .2 .2
+@item b                 @tab bitrate
+FFmpeg @code{b} option is expressed in bits/s, x264 @code{bitrate} in kilobits/s.
+@item bf                @tab bframes
+Maximum number of B-frames.
+@item g                 @tab keyint
+Maximum GOP size.
+@item qmin              @tab qpmin
+@item qmax              @tab qpmax
+@item qdiff             @tab qpstep
+@item qblur             @tab qblur
+@item qcomp             @tab qcomp
+@item refs              @tab ref
+@item sc_threshold      @tab scenecut
+@item trellis           @tab trellis
+@item nr                @tab nr
+Noise reduction.
+@item me_range          @tab merange
+@item me_method         @tab me
+@item subq              @tab subme
+@item b_strategy        @tab b-adapt
+@item keyint_min        @tab keyint-min
+@item coder             @tab cabac
+Set coder to @code{ac} to use CABAC.
+@item cmp               @tab chroma-me
+Set to @code{chroma} to use chroma motion estimation.
+@item threads           @tab threads
+@item thread_type       @tab sliced_threads
+Set to @code{slice} to use sliced threading instead of frame threading.
+@item flags -cgop       @tab open-gop
+Set @code{-cgop} to use recovery points to close GOPs.
+@item rc_init_occupancy @tab vbv-init
+Initial buffer occupancy.
+@end multitable
+
+@subsection Private Options
@table @option
+@item -preset @var{string}
+Set the encoding preset (cf. x264 --fullhelp).
+@item -tune @var{string}
+Tune the encoding params (cf. x264 --fullhelp).
+@item -profile @var{string}
+Set profile restrictions (cf. x264 --fullhelp).
+@item -fastfirstpass @var{integer}
+Use fast settings when encoding first pass.
+@item -crf @var{float}
+Select the quality for constant quality mode.
+@item -crf_max @var{float}
+In CRF mode, prevents VBV from lowering quality beyond this point.
+@item -qp @var{integer}
+Constant quantization parameter rate control method.
+@item -aq-mode @var{integer}
+AQ method

-@item preset @var{preset_name}
-Set the encoding preset.
+Possible values:
+@table @samp
+@item none

-@item tune @var{tune_name}
-Tune the encoding params.
+@item variance
+Variance AQ (complexity mask).
+@item autovariance
+Auto-variance AQ (experimental).
+@end table
+@item -aq-strength @var{float}
+AQ strength, reduces blocking and blurring in flat and textured areas.
+@item -psy @var{integer}
+Use psychovisual optimizations.
+@item -psy-rd @var{string}
+Strength of psychovisual optimization, in <psy-rd>:<psy-trellis> format.
+@item -rc-lookahead @var{integer}
+Number of frames to look ahead for frametype and ratecontrol.
+@item -weightb @var{integer}
+Weighted prediction for B-frames.
+@item -weightp @var{integer}
+Weighted prediction analysis method.

-@item fastfirstpass @var{bool}
-Use fast settings when encoding first pass, default value is 1.
+Possible values:
+@table @samp
+@item none

-@item profile @var{profile_name}
-Set profile restrictions.
+@item simple

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

-@item passlogfile @var{filename}
-Specify filename for 2 pass stats.
-Deprecated in favor of @var{x264opts} (see @var{stats} libx264 option).
+@end table
+@item -ssim @var{integer}
+Calculate and print SSIM stats.
+@item -intra-refresh @var{integer}
+Use Periodic Intra Refresh instead of IDR frames.
+@item -b-bias @var{integer}
+Influences how often B-frames are used.
+@item -b-pyramid @var{integer}
+Keep some B-frames as references.

-@item wpredp @var{wpred_type}
-Specify Weighted prediction for P-frames.
-Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).
+Possible values:
+@table @samp
+@item none
+
+@item strict
+Strictly hierarchical pyramid.
+@item normal
+Non-strict (not Blu-ray compatible).
+@end table
+@item -mixed-refs @var{integer}
+One reference per partition, as opposed to one reference per macroblock.
+@item -8x8dct @var{integer}
+High profile 8x8 transform.
+@item -fast-pskip @var{integer}
+@item -aud @var{integer}
+Use access unit delimiters.
+@item -mbtree @var{integer}
+Use macroblock tree ratecontrol.
+@item -deblock @var{string}
+Loop filter parameters, in <alpha:beta> form.
+@item -cplxblur @var{float}
+Reduce fluctuations in QP (before curve compression).
+@item -partitions @var{string}
+A comma-separated list of partitions to consider, possible values: p8x8, p4x4, b8x8, i8x8, i4x4, none, all.
+@item -direct-pred @var{integer}
+Direct MV prediction mode
+
+Possible values:
+@table @samp
+@item none
+
+@item spatial
+
+@item temporal
+
+@item auto
+
+@end table
+@item -slice-max-size @var{integer}
+Limit the size of each slice in bytes.
+@item -stats @var{string}
+Filename for 2 pass stats.
+@item -nal-hrd @var{integer}
+Signal HRD information (requires vbv-bufsize; cbr not allowed in .mp4).
+
+Possible values:
+@table @samp
+@item none
+
+@item vbr
+
+@item cbr
+
+@end table

@item x264opts @var{options}
-Allow to set any x264 option, see x264 --fullhelp for a list.
+Allow to set any x264 option, see @code{x264 --fullhelp} for a list.

@var{options} is a list of @var{key}=@var{value} couples separated by
-":".
-@end table
+":". In @var{filter} and @var{psy-rd} options that use ":" as a separator
+themselves, use "," instead. They accept it as well since long ago but this
+is kept undocumented for some reason.

 For example to specify libx264 encoding options with @command{ffmpeg}:
@example
@@ -591,4 +1057,94 @@ ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
 For more information about libx264 and the supported options see:
@url{http://www.videolan.org/developers/x264.html}

+@item -x264-params @var{string}
+Override the x264 configuration using a :-separated list of key=value parameters.
+@example
+-x264-params level=30:bframes=0:weightp=0:cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:no-fast-pskip=1:subq=6:8x8dct=0:trellis=0
+@end example
+@end table
+
+Encoding avpresets for common usages are provided so they can be used with the
+general presets system (e.g. passing the @code{-pre} option).
+
+@section png
+
+PNG image encoder.
+
+@subsection Private options
+
+@table @option
+@item dpi @var{integer}
+Set physical density of pixels, in dots per inch, unset by default
+@item dpm @var{integer}
+Set physical density of pixels, in dots per meter, unset by default
+@end table
+
+@section ProRes
+
+Apple ProRes encoder.
+
+FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
+The used encoder can be choosen with the @code{-vcodec} option.
+
+@subsection Private Options for prores-ks
+
+@table @option
+@item profile @var{integer}
+Select the ProRes profile to encode
+@table @samp
+@item proxy
+@item lt
+@item standard
+@item hq
+@item 4444
+@end table
+
+@item quant_mat @var{integer}
+Select quantization matrix.
+@table @samp
+@item auto
+@item default
+@item proxy
+@item lt
+@item standard
+@item hq
+@end table
+If set to @var{auto}, the matrix matching the profile will be picked.
+If not set, the matrix providing the highest quality, @var{default}, will be
+picked.
+
+@item bits_per_mb @var{integer}
+How many bits to allot for coding one macroblock. Different profiles use
+between 200 and 2400 bits per macroblock, the maximum is 8000.
+
+@item mbs_per_slice @var{integer}
+Number of macroblocks in each slice (1-8); the default value (8)
+should be good in almost all situations.
+
+@item vendor @var{string}
+Override the 4-byte vendor ID.
+A custom vendor ID like @var{apl0} would claim the stream was produced by
+the Apple encoder.
+
+@item alpha_bits @var{integer}
+Specify number of bits for alpha component.
+Possible values are @var{0}, @var{8} and @var{16}.
+Use @var{0} to disable alpha plane coding.
+
+@end table
+
+@subsection Speed considerations
+
+In the default mode of operation the encoder has to honor frame constraints
+(i.e. not produc frames with size bigger than requested) while still making
+output picture as good as possible.
+A frame containing a lot of small details is harder to compress and the encoder
+would spend more time searching for appropriate quantizers for each slice.
+
+Setting a higher @option{bits_per_mb} limit will improve the speed.
+
+For the fastest encoding speed set the @option{qscale} parameter (4 is the
+recommended value) and do not set a size constraint.
+
@c man end VIDEO ENCODERS
--- a/doc/eval.texi
+++ b/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
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -7,7 +7,7 @@ FFMPEG_LIBS=    libavdevice                        \
                libswscale                         \
                libavutil                          \

-CFLAGS += -Wall -O2 -g
+CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

@@ -17,6 +17,7 @@ EXAMPLES=       decoding_encoding                  \
                filtering_audio                    \
                metadata                           \
                muxing                             \
+                resampling_audio                   \
                scaling_video                      \

 OBJS=$(addsuffix .o,$(EXAMPLES))
--- a/doc/examples/README
+++ b/doc/examples/README
@@ -0,0 +1,18 @@
+FFmpeg examples README
+----------------------
+
+Both following use cases rely on pkg-config and make, thus make sure
+that you have them installed and working on your system.
+
+
+1) Build the installed examples in a generic read/write user directory
+
+Copy to a read/write user directory and just use "make", it will link
+to the libraries on your system, assuming the PKG_CONFIG_PATH is
+correctly configured.
+
+2) Build the examples in-tree
+
+Assuming you are in the source FFmpeg checkout directory, you need to build
+FFmpeg (no need to make install in any prefix). Then you can go into the
+doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -27,13 +27,14 @@
 * 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/audioconvert.h>
+#include <libavutil/channel_layout.h>
 #include <libavutil/common.h>
 #include <libavutil/imgutils.h>
 #include <libavutil/mathematics.h>
@@ -78,7 +79,7 @@ static int select_channel_layout(AVCodec *codec)
 {
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
-    int best_nb_channells   = 0;
+    int best_nb_channels   = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;
@@ -87,9 +88,9 @@ static int select_channel_layout(AVCodec *codec)
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

-        if (nb_channels > best_nb_channells) {
+        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
-            best_nb_channells = nb_channels;
+            best_nb_channels = nb_channels;
        }
        p++;
    }
@@ -121,6 +122,10 @@ static void audio_encode_example(const char *filename)
    }

    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;
@@ -252,6 +257,10 @@ static void audio_decode_example(const char *outfilename, const char *filename)
    }

    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) {
@@ -346,6 +355,10 @@ static void video_encode_example(const char *filename, int codec_id)
    }

    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;
@@ -356,7 +369,7 @@ static void video_encode_example(const char *filename, int codec_id)
    c->time_base= (AVRational){1,25};
    c->gop_size = 10; /* emit one intra frame every ten frames */
    c->max_b_frames=1;
-    c->pix_fmt = PIX_FMT_YUV420P;
+    c->pix_fmt = AV_PIX_FMT_YUV420P;

    if(codec_id == AV_CODEC_ID_H264)
        av_opt_set(c->priv_data, "preset", "slow", 0);
@@ -475,15 +488,42 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
    fclose(f);
 }

+static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
+                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
+{
+    int len, got_frame;
+    char buf[1024];
+
+    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
+    if (len < 0) {
+        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
+        return len;
+    }
+    if (got_frame) {
+        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder, no need to free it */
+        snprintf(buf, sizeof(buf), outfilename, *frame_count);
+        pgm_save(frame->data[0], frame->linesize[0],
+                 avctx->width, avctx->height, buf);
+        (*frame_count)++;
+    }
+    if (pkt->data) {
+        pkt->size -= len;
+        pkt->data += len;
+    }
+    return 0;
+}
+
 static void video_decode_example(const char *outfilename, const char *filename)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
-    int frame, got_picture, len;
+    int frame_count;
    FILE *f;
-    AVFrame *picture;
+    AVFrame *frame;
    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
-    char buf[1024];
    AVPacket avpkt;

    av_init_packet(&avpkt);
@@ -501,6 +541,11 @@ static void video_decode_example(const char *outfilename, const char *filename)
    }

    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 */

@@ -514,21 +559,19 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }

-    /* the codec gives us the frame size, in samples */
-
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }

-    picture = avcodec_alloc_frame();
-    if (!picture) {
+    frame = avcodec_alloc_frame();
+    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }

-    frame = 0;
+    frame_count = 0;
    for(;;) {
        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
        if (avpkt.size == 0)
@@ -550,26 +593,9 @@ static void video_decode_example(const char *outfilename, const char *filename)
        /* here, we use a stream based decoder (mpeg1video), so we
           feed decoder and see if it could decode a frame */
        avpkt.data = inbuf;
-        while (avpkt.size > 0) {
-            len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
-            if (len < 0) {
-                fprintf(stderr, "Error while decoding frame %d\n", frame);
+        while (avpkt.size > 0)
+            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
                exit(1);
-            }
-            if (got_picture) {
-                printf("Saving frame %3d\n", frame);
-                fflush(stdout);
-
-                /* the picture is allocated by the decoder. no need to
-                   free it */
-                snprintf(buf, sizeof(buf), outfilename, frame);
-                pgm_save(picture->data[0], picture->linesize[0],
-                         c->width, c->height, buf);
-                frame++;
-            }
-            avpkt.size -= len;
-            avpkt.data += len;
-        }
    }

    /* some codecs, such as MPEG, transmit the I and P frame with a
@@ -577,24 +603,13 @@ static void video_decode_example(const char *outfilename, const char *filename)
       chance to get the last frame of the video */
    avpkt.data = NULL;
    avpkt.size = 0;
-    len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
-    if (got_picture) {
-        printf("Saving last frame %3d\n", frame);
-        fflush(stdout);
-
-        /* the picture is allocated by the decoder. no need to
-           free it */
-        snprintf(buf, sizeof(buf), outfilename, frame);
-        pgm_save(picture->data[0], picture->linesize[0],
-                 c->width, c->height, buf);
-        frame++;
-    }
+    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);

    fclose(f);

    avcodec_close(c);
    av_free(c);
-    avcodec_free_frame(&picture);
+    avcodec_free_frame(&frame);
    printf("\n");
 }

@@ -610,8 +625,8 @@ int main(int argc, char **argv)
               "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",
+               "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;
    }
--- a/doc/examples/demuxing.c
+++ b/doc/examples/demuxing.c
@@ -26,6 +26,7 @@
 *
 * 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>
@@ -97,7 +98,7 @@ static int decode_packet(int *got_frame, int 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,
+            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
                                   frame->nb_samples, frame->format, 1);
            if (ret < 0) {
                fprintf(stderr, "Could not allocate audio buffer\n");
@@ -106,13 +107,13 @@ static int decode_packet(int *got_frame, int cached)

            /* 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,
+                av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
                                           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);
+                            frame->nb_samples, av_frame_get_channels(frame), frame->format);

            /* write to rawaudio file */
            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
@@ -208,7 +209,7 @@ int main (int argc, char **argv)
    /* register all formats and codecs */
    av_register_all();

-    /* open input file, and allocated format context */
+    /* 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);
@@ -242,9 +243,6 @@ int main (int argc, char **argv)
        video_dst_bufsize = ret;
    }

-    /* dump input information to stderr */
-    av_dump_format(fmt_ctx, 0, src_filename, 0);
-
    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
        int nb_planes;

@@ -267,6 +265,9 @@ int main (int argc, char **argv)
        }
    }

+    /* 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;
@@ -288,11 +289,13 @@ int main (int argc, char **argv)
    if (video_stream)
        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
    if (audio_stream)
-        printf("Demuxing video from file '%s' into '%s'\n", src_filename, audio_dst_filename);
+        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)
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
        decode_packet(&got_frame, 0);
+        av_free_packet(&pkt);
+    }

    /* flush cached frames */
    pkt.data = NULL;
@@ -313,7 +316,7 @@ int main (int argc, char **argv)
    if (audio_stream) {
        const char *fmt;

-        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt) < 0))
+        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",
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -25,6 +25,7 @@
 /**
 * @file
 * API example for audio decoding and filtering
+ * @example doc/examples/filtering_audio.c
 */

 #include <unistd.h>
@@ -35,8 +36,9 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>

-const char *filter_descr = "aresample=8000,aconvert=s16:mono";
+const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
 const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

 static AVFormatContext *fmt_ctx;
@@ -69,6 +71,7 @@ static int open_input_file(const char *filename)
    }
    audio_stream_index = ret;
    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);

    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -84,11 +87,12 @@ 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");
+    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    AVABufferSinkParams *abuffersink_params;
+    const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

@@ -109,16 +113,34 @@ static int init_filters(const char *filters_descr)
    }

    /* 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);
+                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
        return ret;
    }

+    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
+        return ret;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
+        return ret;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
+        return ret;
+    }
+
    /* Endpoints for the filter graph. */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
@@ -130,7 +152,7 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                    &inputs, &outputs, NULL)) < 0)
        return ret;

@@ -149,11 +171,10 @@ static int init_filters(const char *filters_descr)
    return 0;
 }

-static void print_samplesref(AVFilterBufferRef *samplesref)
+static void print_frame(const AVFrame *frame)
 {
-    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 int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
+    const uint16_t *p     = (uint16_t*)frame->data[0];
    const uint16_t *p_end = p + n;

    while (p < p_end) {
@@ -168,9 +189,14 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame frame;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;

+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
        exit(1);
@@ -187,14 +213,13 @@ int main(int argc, char **argv)

    /* 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);
+            avcodec_get_frame_defaults(frame);
            got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
+            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                continue;
@@ -202,22 +227,20 @@ int main(int argc, char **argv)

            if (got_frame) {
                /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
+                if (av_buffersrc_add_frame_flags(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);
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
                    if(ret < 0)
                        goto end;
-                    if (samplesref) {
-                        print_samplesref(samplesref);
-                        avfilter_unref_bufferp(&samplesref);
-                    }
+                    print_frame(filt_frame);
+                    av_frame_unref(filt_frame);
                }
            }
        }
@@ -228,6 +251,8 @@ end:
    if (dec_ctx)
        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
        char buf[1024];
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -24,6 +24,7 @@
 /**
 * @file
 * API example for decoding and filtering
+ * @example doc/examples/filtering_video.c
 */

 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -84,10 +85,10 @@ static int init_filters(const char *filters_descr)
    char args[512];
    int ret;
    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
+    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
    AVBufferSinkParams *buffersink_params;

    filter_graph = avfilter_graph_alloc();
@@ -128,7 +129,7 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                    &inputs, &outputs, NULL)) < 0)
        return ret;

@@ -137,33 +138,33 @@ static int init_filters(const char *filters_descr)
    return 0;
 }

-static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
+static void display_frame(const AVFrame *frame, AVRational time_base)
 {
    int x, y;
    uint8_t *p0, *p;
    int64_t delay;

-    if (picref->pts != AV_NOPTS_VALUE) {
+    if (frame->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,
+            delay = av_rescale_q(frame->pts - last_pts,
                                 time_base, AV_TIME_BASE_Q);
            if (delay > 0 && delay < 1000000)
                usleep(delay);
        }
-        last_pts = picref->pts;
+        last_pts = frame->pts;
    }

    /* Trivial ASCII grayscale display. */
-    p0 = picref->data[0];
+    p0 = frame->data[0];
    puts("\033c");
-    for (y = 0; y < picref->video->h; y++) {
+    for (y = 0; y < frame->height; y++) {
        p = p0;
-        for (x = 0; x < picref->video->w; x++)
+        for (x = 0; x < frame->width; x++)
            putchar(" .-+#"[*(p++) / 52]);
        putchar('\n');
-        p0 += picref->linesize[0];
+        p0 += frame->linesize[0];
    }
    fflush(stdout);
 }
@@ -172,9 +173,14 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame frame;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;

+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file\n", argv[0]);
        exit(1);
@@ -191,40 +197,36 @@ int main(int argc, char **argv)

    /* 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);
+            avcodec_get_frame_defaults(frame);
            got_frame = 0;
-            ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
+            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);
+                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) {
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
                    break;
                }

-                /* pull filtered pictures from the filtergraph */
+                /* pull filtered frames from the filtergraph */
                while (1) {
-                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                    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);
-                    }
+                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
+                    av_frame_unref(filt_frame);
                }
            }
        }
@@ -235,6 +237,8 @@ end:
    if (dec_ctx)
        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
        char buf[1024];
--- a/doc/examples/metadata.c
+++ b/doc/examples/metadata.c
@@ -23,6 +23,7 @@
 /**
 * @file
 * Shows how the metadata API can be used in application programs.
+ * @example doc/examples/metadata.c
 */

 #include <stdio.h>
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -26,6 +26,7 @@
 *
 * Output a media file in any supported libavformat format.
 * The default codecs are used.
+ * @example doc/examples/muxing.c
 */

 #include <stdlib.h>
@@ -33,38 +34,32 @@
 #include <string.h>
 #include <math.h>

+#include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
+#include <libswresample/swresample.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    PIX_FMT_YUV420P /* default pix_fmt */
+#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 audio output stream
- */
-static AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
-                                  enum AVCodecID codec_id)
+/* Add an output stream. */
+static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
+                            enum AVCodecID codec_id)
 {
    AVCodecContext *c;
    AVStream *st;

-    /* find the audio encoder */
+    /* find the encoder */
    *codec = avcodec_find_encoder(codec_id);
    if (!(*codec)) {
-        fprintf(stderr, "Could not find codec\n");
+        fprintf(stderr, "Could not find encoder for '%s'\n",
+                avcodec_get_name(codec_id));
        exit(1);
    }

@@ -73,32 +68,82 @@ static AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }
-    st->id = 1;
-
+    st->id = oc->nb_streams-1;
    c = st->codec;

-    /* put sample parameters */
-    c->sample_fmt  = AV_SAMPLE_FMT_S16;
-    c->bit_rate    = 64000;
-    c->sample_rate = 44100;
-    c->channels    = 2;
+    switch ((*codec)->type) {
+    case AVMEDIA_TYPE_AUDIO:
+        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
+        c->bit_rate    = 64000;
+        c->sample_rate = 44100;
+        c->channels    = 2;
+        break;

-    // some formats want stream headers to be separate
+    case AVMEDIA_TYPE_VIDEO:
+        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 uint8_t **src_samples_data;
+static int       src_samples_linesize;
+static int       src_nb_samples;
+
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int       dst_samples_linesize;
+int       dst_samples_size;
+
+struct SwrContext *swr_ctx = NULL;
+
 static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
    AVCodecContext *c;
+    int ret;

    c = st->codec;

    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "could not open codec\n");
+    ret = avcodec_open2(c, codec, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
        exit(1);
    }

@@ -108,13 +153,51 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    /* 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);
+    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+        10000 : c->frame_size;
+
+    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+                                             src_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
+        exit(1);
+    }
+
+    /* create resampler context */
+    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
+        swr_ctx = swr_alloc();
+        if (!swr_ctx) {
+            fprintf(stderr, "Could not allocate resampler context\n");
+            exit(1);
+        }
+
+        /* set options */
+        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+
+        /* initialize the resampling context */
+        if ((ret = swr_init(swr_ctx)) < 0) {
+            fprintf(stderr, "Failed to initialize the resampling context\n");
+            exit(1);
+        }
+    }
+
+    /* 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 = src_nb_samples;
+    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                             max_dst_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        exit(1);
+    }
+    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
+                                                  c->sample_fmt, 0);
 }

 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -139,28 +222,62 @@ 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;
+    int got_packet, ret, dst_nb_samples;

    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);
+    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+
+    /* convert samples from native format to destination codec format, using the resampler */
+    if (swr_ctx) {
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_samples_data[0]);
+            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                   dst_nb_samples, c->sample_fmt, 0);
+            if (ret < 0)
+                exit(1);
+            max_dst_nb_samples = dst_nb_samples;
+            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                          c->sample_fmt, 0);
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx,
+                          dst_samples_data, dst_nb_samples,
+                          (const uint8_t **)src_samples_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            exit(1);
+        }
+    } else {
+        dst_samples_data[0] = src_samples_data[0];
+        dst_nb_samples = src_nb_samples;
+    }
+
+    frame->nb_samples = dst_nb_samples;
+    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                             dst_samples_data[0], dst_samples_size, 0);
+
+    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);
+    }

-    avcodec_encode_audio2(c, &pkt, frame, &got_packet);
    if (!got_packet)
        return;

    pkt.stream_index = st->index;

    /* Write the compressed frame to the media file. */
-    if (av_interleaved_write_frame(oc, &pkt) != 0) {
-        fprintf(stderr, "Error while writing audio frame\n");
+    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);
@@ -169,8 +286,8 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
    avcodec_close(st->codec);
-
-    av_free(samples);
+    av_free(src_samples_data[0]);
+    av_free(dst_samples_data[0]);
 }

 /**************************************************************/
@@ -178,64 +295,7 @@ static void close_audio(AVFormatContext *oc, AVStream *st)

 static AVFrame *frame;
 static AVPicture src_picture, dst_picture;
-static uint8_t *video_outbuf;
-static int frame_count, video_outbuf_size;
-
-/* Add a video output stream. */
-static AVStream *add_video_stream(AVFormatContext *oc, AVCodec **codec,
-                                  enum AVCodecID codec_id)
-{
-    AVCodecContext *c;
-    AVStream *st;
-
-    /* find the video encoder */
-    *codec = avcodec_find_encoder(codec_id);
-    if (!(*codec)) {
-        fprintf(stderr, "codec not found\n");
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, *codec);
-    if (!st) {
-        fprintf(stderr, "Could not alloc stream\n");
-        exit(1);
-    }
-
-    c = st->codec;
-
-    avcodec_get_context_defaults3(c, *codec);
-
-    c->codec_id = codec_id;
-
-    /* Put sample parameters. */
-    c->bit_rate = 400000;
-    /* Resolution must be a multiple of two. */
-    c->width    = 352;
-    c->height   = 288;
-    /* 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;
-    }
-    /* Some formats want stream headers to be separate. */
-    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
-
-    return st;
-}
+static int frame_count;

 static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
@@ -243,23 +303,12 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    AVCodecContext *c = st->codec;

    /* open the codec */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    ret = avcodec_open2(c, codec, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
        exit(1);
    }

-    video_outbuf = NULL;
-    if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
-        /* Allocate output buffer. */
-        /* XXX: API change will be done. */
-        /* Buffers passed into lav* can be allocated any way you prefer,
-         * as long as they're aligned enough for the architecture, and
-         * they're freed appropriately (such as using av_free for buffers
-         * allocated with av_malloc). */
-        video_outbuf_size = 200000;
-        video_outbuf      = av_malloc(video_outbuf_size);
-    }
-
    /* allocate and init a re-usable frame */
    frame = avcodec_alloc_frame();
    if (!frame) {
@@ -270,17 +319,18 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    /* 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\n");
+        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 != PIX_FMT_YUV420P) {
-        ret = avpicture_alloc(&src_picture, PIX_FMT_YUV420P, c->width, c->height);
+    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\n");
+            fprintf(stderr, "Could not allocate temporary picture: %s\n",
+                    av_err2str(ret));
            exit(1);
        }
    }
@@ -322,11 +372,11 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
         * frames if using B-frames, so we get the last frames by
         * passing the same picture again. */
    } else {
-        if (c->pix_fmt != PIX_FMT_YUV420P) {
+        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, PIX_FMT_YUV420P,
+                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) {
@@ -345,8 +395,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
    }

    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* Raw video case - the API will change slightly in the near
-         * future for that. */
+        /* Raw video case - directly store the picture in the packet */
        AVPacket pkt;
        av_init_packet(&pkt);

@@ -357,28 +406,19 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)

        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
-        /* encode the image */
-        AVPacket pkt;
-        int got_output;
-
+        AVPacket pkt = { 0 };
+        int got_packet;
        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);
+        /* encode the image */
+        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
        if (ret < 0) {
-            fprintf(stderr, "Error encoding video frame\n");
+            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->pts != AV_NOPTS_VALUE)
-                pkt.pts = av_rescale_q(c->coded_frame->pts,
-                                       c->time_base, st->time_base);
-            if (c->coded_frame->key_frame)
-                pkt.flags |= AV_PKT_FLAG_KEY;

+        if (!ret && got_packet && pkt.size) {
            pkt.stream_index = st->index;

            /* Write the compressed frame to the media file. */
@@ -388,7 +428,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
        }
    }
    if (ret != 0) {
-        fprintf(stderr, "Error while writing video frame\n");
+        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
        exit(1);
    }
    frame_count++;
@@ -400,7 +440,6 @@ static void close_video(AVFormatContext *oc, AVStream *st)
    av_free(src_picture.data[0]);
    av_free(dst_picture.data[0]);
    av_free(frame);
-    av_free(video_outbuf);
 }

 /**************************************************************/
@@ -413,8 +452,8 @@ int main(int argc, char **argv)
    AVFormatContext *oc;
    AVStream *audio_st, *video_st;
    AVCodec *audio_codec, *video_codec;
-    double audio_pts, video_pts;
-    int i;
+    double audio_time, video_time;
+    int ret;

    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();
@@ -422,8 +461,10 @@ int main(int argc, char **argv)
    if (argc != 2) {
        printf("usage: %s output_file\n"
               "API example program to output a media file with libavformat.\n"
+               "This program generates a synthetic audio and video stream, encodes and\n"
+               "muxes them into a file named output_file.\n"
               "The output format is automatically guessed according to the file extension.\n"
-               "Raw images can also be output by using '%%d' in the filename\n"
+               "Raw images can also be output by using '%%d' in the filename.\n"
               "\n", argv[0]);
        return 1;
    }
@@ -445,11 +486,12 @@ int main(int argc, char **argv)
     * and initialize the codecs. */
    video_st = NULL;
    audio_st = NULL;
+
    if (fmt->video_codec != AV_CODEC_ID_NONE) {
-        video_st = add_video_stream(oc, &video_codec, fmt->video_codec);
+        video_st = add_stream(oc, &video_codec, fmt->video_codec);
    }
    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
-        audio_st = add_audio_stream(oc, &audio_codec, fmt->audio_codec);
+        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
    }

    /* Now that all the parameters are set, we can open the audio and
@@ -463,42 +505,39 @@ int main(int argc, char **argv)

    /* open the output file, if needed */
    if (!(fmt->flags & AVFMT_NOFILE)) {
-        if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
-            fprintf(stderr, "Could not open '%s'\n", filename);
+        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. */
-    if (avformat_write_header(oc, NULL) < 0) {
-        fprintf(stderr, "Error occurred when opening output file\n");
+    ret = avformat_write_header(oc, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred when opening output file: %s\n",
+                av_err2str(ret));
        return 1;
    }

-    frame->pts = 0;
+    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;
+        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
+        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 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))
+        if ((!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION))
            break;

        /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
+        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
            write_audio_frame(oc, audio_st);
        } else {
            write_video_frame(oc, video_st);
-            frame->pts++;
+            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
        }
    }

@@ -514,18 +553,12 @@ int main(int argc, char **argv)
    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);
+    avformat_free_context(oc);

    return 0;
 }
--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -0,0 +1,211 @@
+/*
+ * 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 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 = av_samples_alloc_array_and_samples(&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 = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
+                                             dst_nb_samples, dst_sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        goto end;
+    }
+
+    t = 0;
+    do {
+        /* generate synthetic audio */
+        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
+
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
+                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_data[0]);
+            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
+                                   dst_nb_samples, dst_sample_fmt, 1);
+            if (ret < 0)
+                break;
+            max_dst_nb_samples = dst_nb_samples;
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            goto end;
+        }
+        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
+                                                 ret, dst_sample_fmt, 1);
+        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
+        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
+    } while (t < 10);
+
+    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
+        goto end;
+    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
+            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
+
+end:
+    if (dst_file)
+        fclose(dst_file);
+
+    if (src_data)
+        av_freep(&src_data[0]);
+    av_freep(&src_data);
+
+    if (dst_data)
+        av_freep(&dst_data[0]);
+    av_freep(&dst_data);
+
+    swr_free(&swr_ctx);
+    return ret < 0;
+}
--- a/doc/examples/scaling_video.c
+++ b/doc/examples/scaling_video.c
@@ -23,6 +23,7 @@
 /**
 * @file
 * libswscale API use example.
+ * @example doc/examples/scaling_video.c
 */

 #include <libavutil/imgutils.h>
@@ -32,20 +33,18 @@
 static void fill_yuv_image(uint8_t *data[4], int linesize[4],
                           int width, int height, int frame_index)
 {
-    int x, y, i;
-
-    i = 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 + i * 3;
+            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 + i * 2;
-            data[2][y * linesize[2] + x] = 64 + x + i * 5;
+            data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
+            data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
        }
    }
 }
@@ -55,7 +54,7 @@ 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 PixelFormat src_pix_fmt = PIX_FMT_YUV420P, dst_pix_fmt = PIX_FMT_RGB24;
+    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;
@@ -108,7 +107,7 @@ int main(int argc, char **argv)
        goto end;
    }

-    /* buffer is going to be written to rawvideo file, no alignmnet */
+    /* buffer is going to be written to rawvideo file, no alignment */
    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");
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -79,6 +79,17 @@ 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?
@@ -99,7 +110,16 @@ 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...
+@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

 If you have large number of pictures to rename, you can use the
 following command to ease the burden. The command, using the bourne
@@ -122,6 +142,12 @@ 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:
@@ -234,18 +260,18 @@ invoking ffmpeg with several @option{-i} options.
 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.html#amerge, @code{amerge}} filter.
+@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.html#pan, @code{pan}} filter to mix
+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.html#overlay, @code{overlay}} video filter.
+@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.

@end itemize

@@ -254,15 +280,26 @@ the other; it can be done using the

 There are several solutions, depending on the exact circumstances.

-@subsection Concatenating using filters
+@subsection Concatenating using the concat @emph{filter}

-FFmpeg has a @url{http://ffmpeg.org/ffmpeg.html#concat-1, @code{concat}}
-filter designed specifically for that, with examples in the documentation.
+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 at the file level
+@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)
+
+FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
+@code{concat}} protocol designed specifically for that, with examples in the
+documentation.

 A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
-video by merely concatenating the files them.
+video by merely concatenating the files containing 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
@@ -297,7 +334,7 @@ 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 &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -qscale:v 2 -c:v mpeg4 -acodec libmp3lame -q:a 4 output.avi
+ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
@end example

@subsection Concatenating using raw audio and video
@@ -327,7 +364,7 @@ cat temp1.a temp2.a > all.a &
 cat temp1.v temp2.v > all.v &
 ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
       -f yuv4mpegpipe -i all.v \
-       -qscale:v 2 -y output.flv
+       -y output.flv
 rm temp[12].[av] all.[av]
@end example

@@ -356,17 +393,17 @@ Appending @code{:v} to it will do exactly that.
 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
+Most likely, it is through @code{auto-inserted aresample}. 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,
+Then insert the correct @code{aformat} explicitly in the filtergraph,
 specifying the exact format.

@example
-aconvert=s16:stereo:packed
+aformat=sample_fmts=s16:channel_layouts=stereo
@end example

@section Why does FFmpeg not see the subtitles in my VOB file?
@@ -385,6 +422,20 @@ 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?
@@ -408,31 +459,8 @@ with @code{#ifdef}s related to the compiler.

@section Is Microsoft Visual C++ supported?

-No. Microsoft Visual C++ is not compliant to the C99 standard and does
-not - among other things - support the inline assembly used in FFmpeg.
-If you wish to use MSVC++ for your
-project then you can link the MSVC++ code with libav* as long as
-you compile the latter with a working C compiler. For more information, see
-the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
-documentation.
-
-There have been efforts to make FFmpeg compatible with MSVC++ in the
-past. However, they have all been rejected as too intrusive, especially
-since MinGW does the job adequately. None of the core developers
-work with MSVC++ and thus this item is low priority. Should you find
-the silver bullet that solves this problem, feel free to shoot it at us.
-
-We strongly recommend you to move over from MSVC++ to MinGW tools.
-
-@section Can I use FFmpeg or libavcodec under Windows?
-
-Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
-Read the @emph{Windows} section in the FFmpeg documentation to find more
-information.
-
-To get help and instructions for building FFmpeg under Windows, check out
-the FFmpeg Windows Help Forum at
-@url{http://ffmpeg.arrozcru.org/}.
+Yes. Please see the @uref{platform.html, Microsoft Visual C++}
+section in the FFmpeg documentation.

@section Can you add automake, libtool or autoconf support?

@@ -490,8 +518,8 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS

@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?

-You have to implement a URLProtocol, see @file{libavformat/file.c} in
-FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
+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.

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

--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -27,7 +27,7 @@ by visiting this website:

  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
+with their 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
@@ -93,11 +93,11 @@ tests/fate.sh /path/to/fate_config
@end example

  A configuration file template with comments describing the individual
-configuration variables can be found at @file{tests/fate_config.sh.template}.
+configuration variables can be found at @file{doc/fate_config.sh.template}.

@ifhtml
  The mentioned configuration template is also available here:
-@verbatiminclude ../tests/fate_config.sh.template
+@verbatiminclude fate_config.sh.template
@end ifhtml

  Create a configuration that suits your needs, based on the configuration
@@ -131,7 +131,12 @@ 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
+@table @option
+@item RSA
+   d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
+@item ECDSA
+   76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
+@end table

  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
@@ -185,6 +190,8 @@ the synchronisation of the samples directory.
    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}.
+@item GEN
+Set to @var{1} to generate the missing or mismatched references.
@end table

@section Examples
--- a/tests/fate_config.sh.template
+++ b/tests/fate_config.sh.template
@@ -4,16 +4,20 @@ 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
+build_only=     # set to "yes" for a compile-only instance that skips tests

 # the following are optional and map to configure options
 arch=
 cpu=
 cross_prefix=
+as=
 cc=
+ld=
 target_os=
 sysroot=
 target_exec=
 target_path=
+target_samples=
 extra_cflags=
 extra_ldflags=
 extra_libs=
--- a/doc/ffmpeg-bitstream-filters.texi
+++ b/doc/ffmpeg-bitstream-filters.texi
@@ -0,0 +1,45 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFmpeg Bitstream Filters Documentation
+@titlepage
+@center @titlefont{FFmpeg Bitstream Filters Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+This document describes the bitstream filters provided by the
+libavcodec library.
+
+A bitstream filter operates on the encoded stream data, and performs
+bitstream level modifications without performing decoding.
+
+@c man end DESCRIPTION
+
+@include bitstream_filters.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libavcodec.html,libavcodec}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-bitstream-filters
+@settitle FFmpeg bitstream filters
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
@@ -0,0 +1,42 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFmpeg Codecs Documentation
+@titlepage
+@center @titlefont{FFmpeg Codecs Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+This document describes the codecs (decoders and encoders) provided by
+the libavcodec library.
+
+@c man end DESCRIPTION
+
+@include codecs.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-codecs
+@settitle FFmpeg codecs
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-devices.texi
+++ b/doc/ffmpeg-devices.texi
@@ -0,0 +1,42 @@
+\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
+
+@include devices.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libavdevice.html,libavdevice}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-devices
+@settitle FFmpeg devices
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-filters.texi
+++ b/doc/ffmpeg-filters.texi
@@ -0,0 +1,42 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFmpeg Filters Documentation
+@titlepage
+@center @titlefont{FFmpeg Filters Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+This document describes filters, sources, and sinks provided by the
+libavfilter library.
+
+@c man end DESCRIPTION
+
+@include filters.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libavfilter.html,libavfilter}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-filters
+@settitle FFmpeg filters
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-formats.texi
+++ b/doc/ffmpeg-formats.texi
@@ -0,0 +1,42 @@
+\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
+
+@include formats.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libavformat.html,libavformat}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-formats
+@settitle FFmpeg formats
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-protocols.texi
+++ b/doc/ffmpeg-protocols.texi
@@ -0,0 +1,42 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFmpeg Protocols Documentation
+@titlepage
+@center @titlefont{FFmpeg Protocols Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+This document describes the input and output protocols provided by the
+libavformat library.
+
+@c man end DESCRIPTION
+
+@include protocols.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libavformat.html,libavformat}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-protocols
+@settitle FFmpeg protocols
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -0,0 +1,44 @@
+\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
+
+@include resampler.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libswresample.html,libswresample}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-resampler
+@settitle FFmpeg Resampler
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@@ -0,0 +1,43 @@
+\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
+
+@include scaler.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libswscale.html,libswscale}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-scaler
+@settitle FFmpeg video scaling and pixel format converter
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg-utils.texi
+++ b/doc/ffmpeg-utils.texi
@@ -0,0 +1,42 @@
+\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 utils.texi
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{libavutil.html,libavutil}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-utils
+@settitle FFmpeg utilities
+
+@end ignore
+
+@bye
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -11,37 +11,31 @@

@chapter Synopsis

-The generic syntax is:
-
-@example
-@c man begin SYNOPSIS
-ffmpeg [global options] [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
-@c man end
-@end example
+ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ...

@chapter Description
@c man begin DESCRIPTION

-ffmpeg is a very fast video and audio converter that can also grab from
+@command{ffmpeg} is a very fast video and audio converter that can also grab from
 a live audio/video source. It can also convert between arbitrary sample
 rates and resize video on the fly with a high quality polyphase filter.

-ffmpeg reads from an arbitrary number of input "files" (which can be regular
+@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
@code{-i} option, and writes to an arbitrary number of output "files", which are
 specified by a plain output filename. Anything found on the command line which
 cannot be interpreted as an option is considered to be an output filename.

-Each input or output file can in principle contain any number of streams of
-different types (video/audio/subtitle/attachment/data). Allowed number and/or
-types of streams can be limited by the container format. Selecting, which
-streams from which inputs go into output, is done either automatically or with
-the @code{-map} option (see the Stream selection chapter).
+Each input or output file can, in principle, contain any number of streams of
+different types (video/audio/subtitle/attachment/data). The allowed number and/or
+types of streams may be limited by the container format. Selecting which
+streams from which inputs will go into which output is either done automatically
+or with the @code{-map} option (see the Stream selection chapter).

 To refer to input files in options, you must use their indices (0-based). E.g.
-the first input file is @code{0}, the second is @code{1} etc. Similarly, streams
+the first input file is @code{0}, the second is @code{1}, etc. Similarly, streams
 within a file are referred to by their indices. E.g. @code{2:3} refers to the
-fourth stream in the third input file. See also the Stream specifiers chapter.
+fourth stream in the third input file. Also see the Stream specifiers chapter.

 As a general rule, options are applied to the next specified
 file. Therefore, order is important, and you can have the same
@@ -56,9 +50,9 @@ options apply ONLY to the next input or output file and are reset between files.

@itemize
@item
-To set the video bitrate of the output file to 64kbit/s:
+To set the video bitrate of the output file to 64 kbit/s:
@example
-ffmpeg -i input.avi -b:v 64k output.avi
+ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
@end example

@item
@@ -102,14 +96,14 @@ tracking lowest timestamp on any active input stream.
 Encoded packets are then passed to the decoder (unless streamcopy is selected
 for the stream, see further for a description). The decoder produces
 uncompressed frames (raw video/PCM audio/...) which can be processed further by
-filtering (see next section). After filtering the frames are passed to the
-encoder, which encodes them and outputs encoded packets again. Finally those are
+filtering (see next section). After filtering, the frames are passed to the
+encoder, which encodes them and outputs encoded packets. Finally those are
 passed to the muxer, which writes the encoded packets to the output file.

@section Filtering
 Before encoding, @command{ffmpeg} can process raw audio and video frames using
 filters from the libavfilter library. Several chained filters form a filter
-graph.  @command{ffmpeg} distinguishes between two types of filtergraphs -
+graph. @command{ffmpeg} distinguishes between two types of filtergraphs:
 simple and complex.

@subsection Simple filtergraphs
@@ -145,7 +139,7 @@ only sets timestamps and otherwise passes the frames unchanged.

@subsection Complex filtergraphs
 Complex filtergraphs are those which cannot be described as simply a linear
-processing chain applied to one stream. This is the case e.g. when the graph has
+processing chain applied to one stream. This is the case, for example, when the graph has
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:

@@ -170,9 +164,11 @@ input. They can be represented with the following diagram:
@end example

 Complex filtergraphs are configured with the @option{-filter_complex} option.
-Note that this option is global, since a complex filtergraph by its nature
+Note that this option is global, since a complex filtergraph, by its nature,
 cannot be unambiguously associated with a single stream or file.

+The @option{-lavfi} option is equivalent to @option{-filter_complex}.
+
 A trivial example of a complex filtergraph is the @code{overlay} filter, which
 has two video inputs and one video output, containing one video overlaid on top
 of the other. Its audio counterpart is the @code{amix} filter.
@@ -182,7 +178,7 @@ Stream copy is a mode selected by supplying the @code{copy} parameter to the
@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding
 step for the specified stream, so it does only demuxing and muxing. It is useful
 for changing the container format or modifying container-level metadata. The
-diagram above will in this case simplify to this:
+diagram above will, in this case, simplify to this:

@example
 _______              ______________            ________
@@ -194,7 +190,7 @@ diagram above will in this case simplify to this:
@end example

 Since there is no decoding or encoding, it is very fast and there is no quality
-loss. However it might not work in some cases because of many factors. Applying
+loss. However, it might not work in some cases because of many factors. Applying
 filters is obviously also impossible, since filters work on uncompressed data.

@c man end DETAILED DESCRIPTION
@@ -202,14 +198,14 @@ filters is obviously also impossible, since filters work on uncompressed data.
@chapter Stream selection
@c man begin STREAM SELECTION

-By default ffmpeg includes only one stream of each type (video, audio, subtitle)
+By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
 present in the input files and adds them to each output file.  It picks the
-"best" of each based upon the following criteria; for video it is the stream
-with the highest resolution, for audio the stream with the most channels, for
-subtitle it's the first subtitle stream. In the case where several streams of
-the same type rate equally, the lowest numbered stream is chosen.
+"best" of each based upon the following criteria: for video, it is the stream
+with the highest resolution, for audio, it is the stream with the most channels, for
+subtitles, it is the first subtitle stream. In the case where several streams of
+the same type rate equally, the stream with the lowest index is chosen.

-You can disable some of those defaults by using @code{-vn/-an/-sn} options. For
+You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For
 full manual control, use the @code{-map} option, which disables the defaults just
 described.

@@ -226,7 +222,7 @@ described.

@item -f @var{fmt} (@emph{input/output})
 Force input or output file format. The format is normally auto detected for input
-files and guessed from file extension for output files, so this option is not
+files and guessed from the file extension for output files, so this option is not
 needed in most cases.

@item -i @var{filename} (@emph{input})
@@ -236,7 +232,8 @@ input file name
 Overwrite output files without asking.

@item -n (@emph{global})
-Do not overwrite output files but exit if file exists.
+Do not overwrite output files, and exit immediately if a specified
+output file already exists.

@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@@ -262,6 +259,14 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
 Stop writing the output after its duration reaches @var{duration}.
@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.

+-to and -t are mutually exclusive and -t has priority.
+
+@item -to @var{position} (@emph{output})
+Stop writing the output at @var{position}.
+@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
+
+-to and -t are mutually exclusive and -t has priority.
+
@item -fs @var{limit_size} (@emph{output})
 Set the file size limit, expressed in bytes.

@@ -340,18 +345,32 @@ Stop writing to the stream after @var{framecount} frames.
 Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.

-@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream})
-@var{filter_graph} is a description of the filter graph to apply to
-the stream. Use @code{-filters} to show all the available filters
-(including also sources and sinks).
+@anchor{filter_option}
+@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
+Create the filtergraph specified by @var{filtergraph} and use it to
+filter the stream.
+
+@var{filtergraph} is a description of the filtergraph to apply to
+the stream, and must have a single input and a single output of the
+same type of the stream. In the filtergraph, the input is associated
+to the label @code{in}, and the output to the label @code{out}. See
+the ffmpeg-filters manual for more information about the filtergraph
+syntax.
+
+See the @ref{filter_complex_option,,-filter_complex option} if you
+want to create filtergraphs with multiple inputs and/or outputs.
+
+@item -filter_script[:@var{stream_specifier}] @var{filename} (@emph{output,per-stream})
+This option is similar to @option{-filter}, the only difference is that its
+argument is the name of the file from which a filtergraph description is to be
+read.

-See also the @option{-filter_complex} option if you want to create filter graphs
-with multiple inputs and/or outputs.
@item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
 Specify the preset for matching stream(s).

@item -stats (@emph{global})
-Print encoding progress/statistics. On by default.
+Print encoding progress/statistics. It is on by default, to explicitly
+disable it you need to specify @code{-nostats}.

@item -progress @var{url} (@emph{global})
 Send program-friendly progress information to @var{url}.
@@ -401,11 +420,11 @@ will be used.

 E.g. to extract the first attachment to a file named 'out.ttf':
@example
-ffmpeg -dump_attachment:t:0 out.ttf INPUT
+ffmpeg -dump_attachment:t:0 out.ttf -i INPUT
@end example
 To extract all attachments to files determined by the @code{filename} tag:
@example
-ffmpeg -dump_attachment:t "" INPUT
+ffmpeg -dump_attachment:t "" -i INPUT
@end example

 Technical note -- attachments are implemented as codec extradata, so this
@@ -426,8 +445,7 @@ As an input option, ignore any timestamps stored in the file and instead
 generate timestamps assuming constant frame rate @var{fps}.

 As an output option, duplicate or drop input frames to achieve constant output
-frame rate @var{fps} (note that this actually causes the @code{fps} filter to be
-inserted to the end of the corresponding filtergraph).
+frame rate @var{fps}.

@item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
 Set frame size.
@@ -450,31 +468,15 @@ form @var{num}:@var{den}, where @var{num} and @var{den} are the
 numerator and denominator of the aspect ratio. For example "4:3",
 "16:9", "1.3333", and "1.7777" are valid argument values.

-@item -croptop @var{size}
-@item -cropbottom @var{size}
-@item -cropleft @var{size}
-@item -cropright @var{size}
-All the crop options have been removed. Use -vf
-crop=width:height:x:y instead.
-
-@item -padtop @var{size}
-@item -padbottom @var{size}
-@item -padleft @var{size}
-@item -padright @var{size}
-@item -padcolor @var{hex_color}
-All the pad options have been removed. Use -vf
-pad=width:height:x:y:color instead.
+If used together with @option{-vcodec copy}, it will affect the aspect ratio
+stored at container level, but not the aspect ratio stored in encoded
+frames, if it exists.

@item -vn (@emph{output})
 Disable video recording.

@item -vcodec @var{codec} (@emph{output})
 Set the video codec. This is an alias for @code{-codec:v}.
-@item -same_quant
-Use same quantizer as source (implies VBR).
-
-Note that this is NOT SAME QUALITY. Do not use this option unless you know you
-need it.

@item -pass[:@var{stream_specifier}] @var{n} (@emph{output,per-stream})
 Select the pass number (1 or 2). It is used to do two-pass
@@ -498,12 +500,11 @@ stream
@item -vlang @var{code}
 Set the ISO 639 language code (3 letters) of the current video stream.

-@item -vf @var{filter_graph} (@emph{output})
-@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).  This is an alias for @code{-filter:v}.
+@item -vf @var{filtergraph} (@emph{output})
+Create the filtergraph specified by @var{filtergraph} and use it to
+filter the stream.

+This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
@end table

@section Advanced Video Options
@@ -516,7 +517,7 @@ If the selected pixel format can not be selected, ffmpeg will print a
 warning and select the best pixel format supported by the encoder.
 If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error
 if the requested pixel format can not be selected, and automatic conversions
-inside filter graphs are disabled.
+inside filtergraphs are disabled.
 If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format
 as the input (or graph output) and automatic conversions are disabled.

@@ -531,10 +532,6 @@ list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.

-@item -deinterlace
-Deinterlace pictures.
-This option is deprecated since the deinterlacing is very low quality.
-Use the yadif filter with @code{-filter:v yadif}.
@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
@@ -557,12 +554,58 @@ Force video tag/fourcc. This is an alias for @code{-tag:v}.
 Show QP histogram
@item -vbsf @var{bitstream_filter}
 Deprecated see -bsf
+
@item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
+@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
 Force key frames at the specified timestamps, more precisely at the first
 frames after each specified time.
+
+If the argument is prefixed with @code{expr:}, the string @var{expr}
+is interpreted like an expression and is evaluated for each frame. A
+key frame is forced in case the evaluation is non-zero.
+
+If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
+the time of the beginning of all chapters in the file, shifted by
+@var{delta}, expressed as a time in seconds.
 This option can be useful to ensure that a seek point is present at a
 chapter mark or any other designated place in the output file.
-The timestamps must be specified in ascending order.
+
+For example, to insert a key frame at 5 minutes, plus key frames 0.1 second
+before the beginning of every chapter:
+@example
+-force_key_frames 0:05:00,chapters-0.1
+@end example
+
+The expression in @var{expr} can contain the following constants:
+@table @option
+@item n
+the number of current processed frame, starting from 0
+@item n_forced
+the number of forced frames
+@item prev_forced_n
+the number of the previous forced frame, it is @code{NAN} when no
+keyframe was forced yet
+@item prev_forced_t
+the time of the previous forced frame, it is @code{NAN} when no
+keyframe was forced yet
+@item t
+the time of the current processed frame
+@end table
+
+For example to force a key frame every 5 seconds, you can specify:
+@example
+-force_key_frames expr:gte(t,n_forced*5)
+@end example
+
+To force a key frame 5 seconds after the time of the last forced one,
+starting from second 13:
+@example
+-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
+@end example
+
+Note that forcing too many keyframes is very harmful for the lookahead
+algorithms of certain encoders: using fixed-GOP options or similar
+would be more efficient.

@item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
@@ -593,11 +636,12 @@ Set the audio codec. This is an alias for @code{-codec:a}.
@item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.
-@item -af @var{filter_graph} (@emph{output})
-@var{filter_graph} is a description of the filter graph to apply to
-the input audio.
-Use the option "-filters" to show all the available filters (including
-also sources and sinks).  This is an alias for @code{-filter:a}.
+
+@item -af @var{filtergraph} (@emph{output})
+Create the filtergraph specified by @var{filtergraph} and use it to
+filter the stream.
+
+This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
@end table

@section Advanced Audio options:
@@ -607,6 +651,12 @@ also sources and sinks).  This is an alias for @code{-filter:a}.
 Force audio tag/fourcc. This is an alias for @code{-tag:a}.
@item -absf @var{bitstream_filter}
 Deprecated, see -bsf
+@item -guess_layout_max @var{channels} (@emph{input,per-stream})
+If some input channel layout is not known, try to guess only if it
+corresponds to at most the specified number of channels. For example, 2
+tells to @command{ffmpeg} to recognize 1 channel as mono and 2 channels as
+stereo but not 6 channels as 5.1. The default is to always try to guess. Use
+0 to disable all guessing.
@end table

@section Subtitle options:
@@ -639,6 +689,9 @@ Note that this option will delay the output of all data until the next
 subtitle packet is decoded: it may increase memory consumption and latency a
 lot.

+@item -canvas_size @var{size}
+Set the size of the canvas used to render subtitles.
+
@end table

@section Advanced options
@@ -801,39 +854,7 @@ Copy chapters from input file with index @var{input_file_index} to the next
 output file. If no chapter mapping is specified, then chapters are copied from
 the first input file with at least one chapter. Use a negative file index to
 disable any chapter copying.
-@item -debug @var{category}
-Print specific debug info.
-@var{category} is a number or a string containing one of the following values:
-@table @samp
-@item bitstream
-@item buffers
-picture buffer allocations
-@item bugs
-@item dct_coeff
-@item er
-error recognition
-@item mb_type
-macroblock (MB) type
-@item mmco
-memory management control operations (H.264)
-@item mv
-motion vector
-@item pict
-picture info
-@item pts
-@item qp
-per-block quantization parameter (QP)
-@item rc
-rate control
-@item skip
-@item startcode
-@item thread_ops
-threading operations
-@item vis_mb_type
-visualize block types
-@item vis_qp
-visualize quantization parameter (QP), lower QP are tinted greener
-@end table
+
@item -benchmark (@emph{global})
 Show benchmarking information at the end of an encode.
 Shows CPU time used and maximum memory consumption.
@@ -875,7 +896,7 @@ Newly added values will have to be specified as strings always.
 Each frame is passed with its timestamp from the demuxer to the muxer.
@item 1, cfr
 Frames will be duplicated and dropped to achieve exactly the requested
-constant framerate.
+constant frame rate.
@item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
@@ -887,6 +908,10 @@ Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
@end table

+Note that the timestamps may be further modified by the muxer, after this.
+For example, in the case that the format option @option{avoid_negative_ts}
+is enabled.
+
 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.
@@ -896,9 +921,23 @@ 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.
-This option has been deprecated. Use the @code{asyncts} audio filter instead.
+
+Note that the timestamps may be further modified by the muxer, after this.
+For example, in the case that the format option @option{avoid_negative_ts}
+is enabled.
+
+This option has been deprecated. Use the @code{aresample} audio filter instead.
+
@item -copyts
-Copy timestamps from input to output.
+Do not process input timestamps, but keep their values without trying
+to sanitize them. In particular, do not remove the initial start time
+offset value.
+
+Note that, depending on the @option{vsync} option or on specific muxer
+processing (e.g. in case the format option @option{avoid_negative_ts}
+is enabled) the output timestamps may mismatch with the input
+timestamps even when this option is selected.
+
@item -copytb @var{mode}
 Specify how to set the encoder timebase when stream copying.  @var{mode} is an
 integer numeric value, and can assume one of the following values:
@@ -944,7 +983,7 @@ ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
@end example

@item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
-Set bitstream filters for matching streams. @var{bistream_filters} is
+Set bitstream filters for matching streams. @var{bitstream_filters} is
 a comma-separated list of bitstream filters. Use the @code{-bsfs} option
 to get the list of bitstream filters.
@example
@@ -964,11 +1003,13 @@ Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
 ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
@end example

+@anchor{filter_complex_option}
@item -filter_complex @var{filtergraph} (@emph{global})
-Define a complex filter graph, i.e. one with arbitrary number of inputs and/or
+Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
 outputs. For simple graphs -- those with one input and one output of the same
 type -- see the @option{-filter} options. @var{filtergraph} is a description of
-the filter graph, as described in @ref{Filtergraph syntax}.
+the filtergraph, as described in the ``Filtergraph syntax'' section of the
+ffmpeg-filters manual.

 Input link labels must refer to input streams using the
@code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map}
@@ -1007,13 +1048,31 @@ ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv

 To generate 5 seconds of pure red video using lavfi @code{color} source:
@example
-ffmpeg -filter_complex 'color=red' -t 5 out.mkv
+ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
@end example
+
+@item -lavfi @var{filtergraph} (@emph{global})
+Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
+outputs. Equivalent to @option{-filter_complex}.
+
+@item -filter_complex_script @var{filename} (@emph{global})
+This option is similar to @option{-filter_complex}, the only difference is that
+its argument is the name of the file from which a complex filtergraph
+description is to be read.
+
+@item -override_ffserver (@emph{global})
+Overrides the input specifications from ffserver. Using this option you can
+map any input stream to ffserver and control many aspects of the encoding from
+ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
+ffserver.
+The option is intended for cases where features are needed that cannot be
+specified to ffserver but can be to ffmpeg.
+
@end table

 As a special exception, you can use a bitmap subtitle stream as input: it
 will be converted into a video with the same size as the largest video in
-the file, or 720×576 if no video is present. Note that this is an
+the file, or 720x576 if no video is present. Note that this is an
 experimental and temporary solution. It will be removed once libavfilter has
 proper support for subtitles.

@@ -1289,7 +1348,7 @@ ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
 You can put many streams of the same type in the output:

@example
-ffmpeg -i test1.avi -i test2.avi -map 0.3 -map 0.2 -map 0.1 -map 0.0 -c copy test12.nut
+ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
@end example

 The resulting output file @file{test12.avi} will contain first four streams from
@@ -1311,32 +1370,74 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end itemize
@c man end EXAMPLES

-@include syntax.texi
-@include eval.texi
-@include decoders.texi
-@include encoders.texi
-@include demuxers.texi
-@include muxers.texi
-@include indevs.texi
-@include outdevs.texi
-@include protocols.texi
+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
@include filters.texi
-@include metadata.texi
+@end ifset
+@end ifset
+
+@chapter See Also
+
+@ifhtml
+@ifset config-all
+@url{ffmpeg.html,ffmpeg}
+@end ifset
+@ifset config-not-all
+@url{ffmpeg-all.html,ffmpeg-all},
+@end ifset
+@url{ffplay.html,ffplay}, @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.html,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
+@ifset config-all
+ffmpeg(1),
+@end ifset
+@ifset config-not-all
+ffmpeg-all(1),
+@end ifset
+ffplay(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 ffmpeg
@settitle ffmpeg video converter

-@c man begin SEEALSO
-ffplay(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
-@c man end
-
-@c man begin AUTHORS
-See git history
-@c man end
-
@end ignore

@bye
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -11,11 +11,7 @@

@chapter Synopsis

-@example
-@c man begin SYNOPSIS
-ffplay [options] [@file{input_file}]
-@c man end
-@end example
+ffplay [@var{options}] [@file{input_file}]

@chapter Description
@c man begin DESCRIPTION
@@ -77,11 +73,22 @@ Default value is "video", if video is not present or cannot be played
 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.
+@item -vf @var{filtergraph}
+Create the filtergraph specified by @var{filtergraph} and use it to
+filter the video stream.
+
+@var{filtergraph} is a description of the filtergraph to apply to
+the stream, and must have a single video input and a single video
+output. In the filtergraph, the input is associated to the label
+@code{in}, and the output to the label @code{out}. See the
+ffmpeg-filters manual for more information about the filtergraph
+syntax.
+
+@item -af @var{filtergraph}
+@var{filtergraph} is a description of the filtergraph to apply to
+the input audio.
 Use the option "-filters" to show all the available filters (including
-also sources and sinks).
+sources and sinks).

@item -i @var{input_file}
 Read @var{input_file}.
@@ -92,9 +99,13 @@ Read @var{input_file}.
@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.
+Print several playback statistics, in particular show the stream
+duration, the codec parameters, the current position in the stream and
+the audio/video synchronisation drift. It is on by default, to
+explicitly disable it you need to specify @code{-nostats}.
+
@item -bug
 Work around bugs.
@item -fast
@@ -134,8 +145,20 @@ Exit when video is done playing.
 Exit if any key is pressed.
@item -exitonmousedown
 Exit if any mouse button is pressed.
-@item -codec:@var{stream_type}
-Force a specific decoder implementation
+
+@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
@@ -178,29 +201,74 @@ Seek to percentage in file corresponding to fraction of width.

@c man end

-@include syntax.texi
-@include eval.texi
-@include decoders.texi
-@include demuxers.texi
-@include muxers.texi
-@include indevs.texi
-@include outdevs.texi
+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
@include filters.texi
+@end ifset
+@end ifset
+
+@chapter See Also
+
+@ifhtml
+@ifset config-all
+@url{ffplay.html,ffplay},
+@end ifset
+@ifset config-not-all
+@url{ffplay-all.html,ffmpeg-all},
+@end ifset
+@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.html,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
+@ifset config-all
+ffplay(1),
+@end ifset
+@ifset config-not-all
+ffplay-all(1),
+@end ifset
+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), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
-@c man end
-
-@c man begin AUTHORS
-The FFmpeg developers
-@c man end
-
@end ignore

@bye
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -11,13 +11,7 @@

@chapter Synopsis

-The generic syntax is:
-
-@example
-@c man begin SYNOPSIS
-ffprobe [options] [@file{input_file}]
-@c man end
-@end example
+ffprobe [@var{options}] [@file{input_file}]

@chapter Description
@c man begin DESCRIPTION
@@ -45,6 +39,10 @@ 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.

@@ -94,6 +92,25 @@ For example for printing the output in JSON format, specify:
 For more details on the available output printing formats, see the
 Writers section below.

+@item -sections
+Print sections structure and section information, and exit. The output
+is not meant to be parsed by a machine.
+
+@item -select_streams @var{stream_specifier}
+Select only the streams specified by @var{stream_specifier}. This
+option affects only the options related to streams
+(e.g. @code{show_streams}, @code{show_packets}, etc.).
+
+For example to show only audio streams, you can use the command:
+@example
+ffprobe -show_streams -select_streams a INPUT
+@end example
+
+To show only video packets belonging to the video stream with index 1:
+@example
+ffprobe -show_packets -select_streams v:1 INPUT
+@end example
+
@item -show_data
 Show payload data, as an hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@@ -118,6 +135,59 @@ 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.
@@ -139,6 +209,11 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".

+@item -show_chapters
+Show information about chapters stored in the format.
+
+Each chapter is printed within a dedicated section with name "CHAPTER".
+
@item -count_frames
 Count the number of frames per stream and report it in the
 corresponding stream section.
@@ -186,8 +261,9 @@ Read @var{input_file}.
 A writer defines the output format adopted by @command{ffprobe}, and will be
 used for printing all the parts of the output.

-A writer may accept one or more arguments, which specify the options to
-adopt.
+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.

@@ -206,9 +282,6 @@ keyN=valN
 Metadata tags are printed as a line in the corresponding FORMAT or
 STREAM section, and are prefixed by the string "TAG:".

-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
 A description of the accepted options follows.

@table @option
@@ -238,9 +311,6 @@ Metadata tags are printed in the corresponding "format" or "stream"
 section. A metadata tag key, if printed, is prefixed by the string
 "tag:".

-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
 The description of the accepted options follows.

@table @option
@@ -292,9 +362,6 @@ A free-form output where each line contains an explicit key=value, such as
 directly embedded in sh scripts as long as the separator character is an
 alphanumeric character or an underscore (see @var{sep_char} option).

-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
 The description of the accepted options follows.

@table @option
@@ -357,9 +424,6 @@ JSON based format.

 Each section is printed using JSON notation.

-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
 The description of the accepted options follows.

@table @option
@@ -386,9 +450,6 @@ Note that the output issued will be compliant to the
 (@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@option{sexagesimal} etc.) are specified.

-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
 The description of the accepted options follows.

@table @option
@@ -431,25 +492,74 @@ DV, GXF and AVI timecodes are available in format metadata
@end itemize
@c man end TIMECODE

-@include syntax.texi
-@include decoders.texi
-@include demuxers.texi
+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
@include protocols.texi
-@include indevs.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
+@chapter See Also
+
+@ifhtml
+@ifset config-all
+@url{ffprobe.html,ffprobe},
+@end ifset
+@ifset config-not-all
+@url{ffprobe-all.html,ffprobe-all},
+@end ifset
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @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.html,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
+@ifset config-all
+ffprobe(1),
+@end ifset
+@ifset config-not-all
+ffprobe-all(1),
+@end ifset
+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

-@c man begin SEEALSO
-ffmpeg(1), ffplay(1), ffserver(1) and the FFmpeg HTML documentation
-@c man end
-
-@c man begin AUTHORS
-The FFmpeg developers
-@c man end
-
@end ignore

@bye
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -11,6 +11,7 @@
            <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="chapters" type="ffprobe:chaptersType" 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" />
@@ -59,6 +60,7 @@
      <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"/>
@@ -77,7 +79,6 @@
      <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">
@@ -86,9 +87,24 @@
        </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"/>
@@ -100,8 +116,6 @@
      <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" />
-      <xsd:attribute name="default"          type="xsd:int" use="required"/>
-      <xsd:attribute name="forced"           type="xsd:int" use="required"/>

      <!-- video attributes -->
      <xsd:attribute name="width"                type="xsd:int"/>
@@ -112,7 +126,6 @@
      <xsd:attribute name="pix_fmt"              type="xsd:string"/>
      <xsd:attribute name="level"                type="xsd:int"/>
      <xsd:attribute name="timecode"             type="xsd:string"/>
-      <xsd:attribute name="attached_pic"         type="xsd:int"/>

      <!-- audio attributes -->
      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
@@ -169,12 +182,32 @@
      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
    </xsd:complexType>

+    <xsd:complexType name="chaptersType">
+      <xsd:sequence>
+        <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+    </xsd:complexType>
+
+    <xsd:complexType name="chapterType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+
+      <xsd:attribute name="id"         type="xsd:int" use="required"/>
+      <xsd:attribute name="time_base"  type="xsd:string" use="required"/>
+      <xsd:attribute name="start"      type="xsd:int" use="required"/>
+      <xsd:attribute name="start_time" type="xsd:float"/>
+      <xsd:attribute name="end"        type="xsd:int" use="required"/>
+      <xsd:attribute name="end_time"   type="xsd:float" 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">
--- a/doc/ffserver.conf
+++ b/doc/ffserver.conf
@@ -25,10 +25,6 @@ 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
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -9,52 +9,35 @@

@contents

-@chapter Synopsys
+@chapter Synopsis

-The generic syntax is:
-
-@example
-@c man begin SYNOPSIS
-ffserver [options]
-@c man end
-@end example
+ffserver [@var{options}]

@chapter Description
@c man begin DESCRIPTION

-ffserver is a streaming server for both audio and video. It supports
+@command{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
+@file{ffserver.conf}).

-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).
+@command{ffserver} receives prerecorded files or FFM streams from some
+@command{ffmpeg} instance as input, then streams them over
+RTP/RTSP/HTTP.

-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.
+An @command{ffserver} instance will listen on some port as specified
+in the configuration file. You can launch one or more instances of
+@command{ffmpeg} and send one or more FFM streams to the port where
+ffserver is expecting to receive them. Alternately, you can make
+@command{ffserver} launch such @command{ffmpeg} instances at startup.

-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
-information.
-
-@section How does it work?
-
-ffserver receives prerecorded files or FFM streams from some ffmpeg
-instance as input, then streams them over RTP/RTSP/HTTP.
-
-An ffserver instance will listen on some port as specified in the
-configuration file. You can launch one or more instances of ffmpeg and
-send one or more FFM streams to the port where ffserver is expecting
-to receive them. Alternately, you can make ffserver launch such ffmpeg
-instances at startup.
-
-Input streams are called feeds, and each one is specified by a <Feed>
-section in the configuration file.
+Input streams are called feeds, and each one is specified by a
+@code{<Feed>} section in the configuration file.

 For each feed you can have different output streams in various
-formats, each one specified by a <Stream> section in the configuration
-file.
+formats, each one specified by a @code{<Stream>} section in the
+configuration file.

@section Status stream

@@ -90,14 +73,6 @@ web server can be used to serve up the files just as well.
 It can stream prerecorded video from .ffm files, though it is somewhat tricky
 to make it work correctly.

-@section What do I need?
-
-I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I'm
-using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
-I needed some special drivers for my motherboard-based sound card.]
-
-I understand that FreeBSD systems work just fine as well.
-
@section How do I make it work?

 First, build the kit. It *really* helps to have installed LAME first. Then when
@@ -238,6 +213,19 @@ You use this by adding the ?date= to the end of the URL for the stream.
 For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end

+@section What is FFM, FFM2
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
@chapter Options
@c man begin OPTIONS

@@ -254,26 +242,79 @@ within the various <Stream> sections. Since ffserver will not launch
 any ffmpeg instances, you will have to launch them manually.
@item -d
 Enable debug mode. This option increases log verbosity, directs log
-messages to stdout and causes ffserver to run in the foreground
-rather than as a daemon.
+messages to stdout.
@end table
@c man end

+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
+@chapter See Also
+
+@ifhtml
+@ifset config-all
+@url{ffserver.html,ffserver},
+@end ifset
+@ifset config-not-all
+@url{ffserver-all.html,ffserver-all},
+@end ifset
+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.html,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
+@ifset config-all
+ffserver(1),
+@end ifset
+@ifset config-not-all
+ffserver-all(1),
+@end ifset
+the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
+ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
+ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
+@end ifnothtml
+
+@include authors.texi
+
@ignore

@setfilename ffserver
@settitle ffserver video server

-@c man begin SEEALSO
-
-ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffserver.conf}
-example and the FFmpeg HTML documentation
-@c man end
-
-@c man begin AUTHORS
-The FFmpeg developers
-@c man end
-
@end ignore

@bye
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -15,13 +15,13 @@ Format negotiation
  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).
+  channel layout, sample format (the sample packing is implied by the sample
+  format) and sample rate.

  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
+  supported at each end 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.

@@ -29,6 +29,11 @@ Format negotiation
  same format amongst a supported list, all it has to do is use a reference
  to the same list of formats.

+  query_formats can leave some formats unset and return AVERROR(EAGAIN) to
+  cause the negotiation mechanism to try again later. That can be used by
+  filters with complex requirements to use the format negotiated on one link
+  to set the formats supported on another.
+

 Buffer references ownership and permissions
 ===========================================
@@ -52,7 +57,7 @@ Buffer references ownership and permissions
    point to only a part of a video buffer.

    A reference is usually obtained as input to the start_frame or
-    filter_samples method or requested using the ff_get_video_buffer 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.
@@ -68,15 +73,15 @@ Buffer references ownership and permissions

    Here are the (fairly obvious) rules for reference ownership:

-    * A reference received by the start_frame or filter_samples method
-      belong to the corresponding filter.
+    * A reference received by the filter_frame method (or its start_frame
+      deprecated version) belongs 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_samples is given
-      away and must no longer be used.
+    * A reference passed to ff_filter_frame (or the deprecated
+      ff_start_frame) is given away and must no longer be used.

    * A reference created with avfilter_ref_buffer belongs to the code that
      created it.
@@ -90,27 +95,11 @@ Buffer references ownership and permissions
  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_samples 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_samples, 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_samples, 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.
+    The AVFilterLink structure has a few AVFilterBufferRef fields. The
+    cur_buf and out_buf were used with the deprecated
+    start_frame/draw_slice/end_frame API and should no longer be used.
+    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
+    and must not be accessed by filters.

  Reference permissions
  ---------------------
@@ -119,8 +108,10 @@ Buffer references ownership and permissions
    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.
+    For video filters that implement the deprecated
+    start_frame/draw_slice/end_frame API, 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:

@@ -179,9 +170,9 @@ Buffer references ownership and permissions
      with the WRITE permission.

    * Filters that intend to keep a reference after the filtering process
-      is finished (after end_frame or filter_samples returns) must have the
-      PRESERVE permission on it and remove the WRITE permission if they
-      create a new reference to give it away.
+      is finished (after 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
@@ -198,11 +189,11 @@ Frame scheduling
  Simple filters that output one frame for each input frame should not have
  to worry about it.

-  start_frame / filter_samples
-  ----------------------------
+  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.
+    This method is called when a frame is pushed to the filter's input. It
+    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.
@@ -213,7 +204,7 @@ Frame scheduling
    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
+    (Example: frame rate-doubling filter: filter_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.)

@@ -233,8 +224,8 @@ Frame scheduling

    This method is called when a frame is wanted on an output.

-    For an input, it should directly call start_frame or filter_samples on
-    the corresponding output.
+    For an input, it should directly call 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
@@ -255,7 +246,7 @@ Frame scheduling
        }
        while (!frame_pushed) {
            input = input_where_a_frame_is_most_needed();
-            ret = avfilter_request_frame(input);
+            ret = ff_request_frame(input);
            if (ret == AVERROR_EOF) {
                process_eof_on_input();
            } else if (ret < 0) {
@@ -266,4 +257,14 @@ Frame scheduling

    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_samples method will be called and do the work.
+    the filter_frame method will be called and do the work.
+
+Legacy API
+==========
+
+  Until libavfilter 3.23, the filter_frame method was split:
+
+  - for video filters, it was made of start_frame, draw_slice (that could be
+    called several times on distinct parts of the frame) and end_frame;
+
+  - for audio filters, it was called filter_samples.
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/doc/formats.texi
@@ -0,0 +1,146 @@
+@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 probe the input. A
+higher value will allow to detect more accurate information, but will
+increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
+
+@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 non-negative. A value of 1 enables shifting,
+a value of 0 disables it, the default value of -1 enables shifting
+when required by the target format.
+
+When shifting is enabled, all output timestamps are shifted by the
+same amount. Audio, video, and subtitles desynching and relative
+timestamp differences are preserved compared to how they would have
+been without shifting.
+
+Also note that this affects only leading negative timestamps, and not
+non-monotonic negative timestamps.
+
+@item flush_packets @var{integer} (@emph{output})
+Flush the underlying I/O stream after each packet. Default 1 enables it, and
+has the effect of reducing the latency; 0 disables it and may slightly
+increase performance in some cases.
+@end table
+
+@c man end FORMAT OPTIONS
+
+@include demuxers.texi
+@include muxers.texi
+@include metadata.texi
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -24,7 +24,7 @@ instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
@file{./configure}.


-@section OpenCORE and VisualOn libraries
+@section OpenCORE, VisualOn, and Fraunhofer libraries

 Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
 libraries provide encoders for a number of audio codecs.
@@ -32,9 +32,14 @@ libraries provide encoders for a number of audio codecs.
@float NOTE
 OpenCORE and VisualOn libraries are under the Apache License 2.0
 (see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
-incompatible with the LGPL version 2.1 and GPL version 2. You have to
+incompatible to the LGPL version 2.1 and GPL version 2. You have to
 upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
-GPL components, GPL version 3) to use it.
+GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
+order to use it.
+
+The Fraunhofer AAC library is licensed under a license incompatible to the GPL
+and is not known to be compatible to the LGPL. Therefore, you have to pass
+@code{--enable-nonfree} to configure to use it.
@end float

@subsection OpenCORE AMR
@@ -95,6 +100,14 @@ Go to @url{http://www.webmproject.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libvpx} to configure to
 enable it.

+@section libwavpack
+
+FFmpeg can make use of the libwavpack library for WavPack encoding.
+
+Go to @url{http://www.wavpack.com/} and follow the instructions for
+installing the library. Then pass @code{--enable-libwavpack} to configure to
+enable it.
+
@section x264

 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -147,9 +160,17 @@ library:
    @tab Multimedia format used in game Heart Of Darkness.
@item Apple HTTP Live Streaming @tab   @tab X
@item Artworx Data Format       @tab   @tab X
+@item ADP                       @tab   @tab X
+    @tab Audio format used on the Nintendo Gamecube.
+@item AFC                       @tab   @tab X
+    @tab Audio format used on the Nintendo Gamecube.
@item ASF                       @tab X @tab X
+@item AST                       @tab X @tab X
+    @tab Audio format used on the Nintendo Wii.
@item AVI                       @tab X @tab X
@item AVISynth                  @tab   @tab X
+@item AVR                       @tab   @tab X
+    @tab Audio format used on Mac.
@item AVS                       @tab   @tab X
    @tab Multimedia format used by the Creature Shock game.
@item Beam Software SIFF        @tab   @tab X
@@ -163,6 +184,8 @@ library:
    @tab Used in Z and Z95 games.
@item Brute Force & Ignorance   @tab   @tab X
    @tab Used in the game Flash Traffic: City of Angels.
+@item BRSTM                     @tab   @tab X
+    @tab Audio format used on the Nintendo Wii.
@item BWF                       @tab X @tab X
@item CRI ADX                   @tab X @tab X
    @tab Audio-only format used in console video games.
@@ -193,6 +216,7 @@ library:
@item Electronic Arts cdata  @tab    @tab X
@item Electronic Arts Multimedia  @tab    @tab X
    @tab Used in various EA games; files have extensions like WVE and UV2.
+@item Ensoniq Paris Audio File  @tab   @tab X
@item FFM (FFserver live feed)  @tab X @tab X
@item Flash (SWF)               @tab X @tab X
@item Flash 9 (AVM2)            @tab X @tab X
@@ -207,7 +231,7 @@ library:
@item G.723.1                   @tab X @tab X
@item G.729 BIT                 @tab X @tab X
@item G.729 raw                 @tab   @tab X
-@item GIF Animation             @tab X @tab
+@item GIF Animation             @tab X @tab X
@item GXF                       @tab X @tab X
    @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
         playout servers.
@@ -227,11 +251,13 @@ library:
    @tab A format generated by IndigoVision 8000 video server.
@item IVF (On2)                 @tab X @tab X
    @tab A format used by libvpx
+@item IRCAM                     @tab X @tab X
@item LATM                      @tab X @tab X
@item LMLM4                     @tab   @tab X
    @tab Used by Linux Media Labs MPEG-4 PCI boards
@item LOAS                      @tab   @tab X
    @tab contains LATM multiplexed AAC audio
+@item LVF                       @tab   @tab X
@item LXF                       @tab   @tab X
    @tab VR native stream format, used by Leitch/Harris' video servers.
@item Matroska                  @tab X @tab X
@@ -242,6 +268,8 @@ library:
    @tab Used in Sim City 3000; file extension .xa.
@item MD Studio                 @tab   @tab X
@item Metal Gear Solid: The Twin Snakes @tab @tab X
+@item Megalux Frame             @tab   @tab X
+    @tab Used by Megalux Ultimate Paint
@item Mobotix .mxg              @tab   @tab X
@item Monkey's Audio            @tab   @tab X
@item Motion Pixels MVI         @tab   @tab X
@@ -269,6 +297,7 @@ library:
    @tab SMPTE 386M, D-10/IMX Mapping.
@item NC camera feed            @tab   @tab X
    @tab NC (AVIP NC4600) camera streams
+@item NIST SPeech HEader REsources @tab   @tab X
@item NTT TwinVQ (VQF)          @tab   @tab X
    @tab Nippon Telegraph and Telephone Corporation TwinVQ.
@item Nullsoft Streaming Video  @tab   @tab X
@@ -277,6 +306,7 @@ library:
    @tab NUT Open Container Format
@item Ogg                       @tab X @tab X
@item Playstation Portable PMP  @tab   @tab X
+@item Portable Voice Format     @tab   @tab X
@item TechnoTrend PVA           @tab   @tab X
    @tab Used by TechnoTrend DVB PCI boards.
@item QCP                       @tab   @tab X
@@ -287,6 +317,7 @@ library:
@item raw Dirac                 @tab X @tab X
@item raw DNxHD                 @tab X @tab X
@item raw DTS                   @tab X @tab X
+@item raw DTS-HD                @tab   @tab X
@item raw E-AC-3                @tab X @tab X
@item raw FLAC                  @tab X @tab X
@item raw GSM                   @tab   @tab X
@@ -304,8 +335,9 @@ library:
@item raw video                 @tab X @tab X
@item raw id RoQ                @tab X @tab
@item raw Shorten               @tab   @tab X
+@item raw TAK                   @tab   @tab X
@item raw TrueHD                @tab X @tab X
-@item raw VC-1                  @tab   @tab X
+@item raw VC-1                  @tab X @tab X
@item raw PCM A-law             @tab X @tab X
@item raw PCM mu-law            @tab X @tab X
@item raw PCM signed 8 bit      @tab X @tab X
@@ -331,11 +363,13 @@ library:
    @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.
@item RealMedia                 @tab X @tab X
@item Redirector                @tab   @tab X
+@item RedSpark                  @tab   @tab X
@item Renderware TeXture Dictionary @tab   @tab X
@item RL2                       @tab   @tab X
    @tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie               @tab   @tab X
@item Lego Mindstorms RSO       @tab X @tab X
+@item RSD                       @tab   @tab X
@item RTMP                      @tab X @tab X
    @tab Output is performed by publishing stream to RTMP server
@item RTP                       @tab X @tab X
@@ -345,6 +379,7 @@ library:
@item SDP                       @tab   @tab X
@item Sega FILM/CPK             @tab   @tab X
    @tab Used in many Sega Saturn console games.
+@item Silicon Graphics Movie    @tab   @tab X
@item Sierra SOL                @tab   @tab X
    @tab .sol files used in Sierra Online games.
@item Sierra VMD                @tab   @tab X
@@ -358,7 +393,7 @@ library:
@item Sony OpenMG (OMA)         @tab X @tab X
    @tab Audio format used in Sony Sonic Stage and Sony Vegas.
@item Sony PlayStation STR      @tab   @tab X
-@item Sony Wave64 (W64)         @tab   @tab X
+@item Sony Wave64 (W64)         @tab X @tab X
@item SoX native format         @tab X @tab X
@item SUN AU format             @tab X @tab X
@item Text files                @tab   @tab X
@@ -368,6 +403,7 @@ library:
    @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio                @tab   @tab X
@item VC-1 test bitstream       @tab X @tab X
+@item Vivo                      @tab   @tab X
@item WAV                       @tab X @tab X
@item WavPack                   @tab X @tab X
@item WebM                      @tab X @tab X
@@ -399,9 +435,10 @@ following image formats are supported:
@item .Y.U.V       @tab X @tab X
    @tab one raw file per component
@item animated GIF @tab X @tab X
-    @tab Only uncompressed GIFs are generated.
@item BMP          @tab X @tab X
    @tab Microsoft BMP image
+@item PIX          @tab   @tab X
+    @tab PIX is an image format used in the Argonaut BRender engine.
@item DPX          @tab X @tab X
    @tab Digital Picture Exchange
@item EXR          @tab   @tab X
@@ -439,6 +476,8 @@ following image formats are supported:
    @tab Targa (.TGA) image format
@item XBM  @tab X @tab X
    @tab X BitMap image format
+@item XFace @tab X @tab X
+    @tab X-Face image format
@item XWD  @tab X @tab X
    @tab X Window Dump image format
@end multitable
@@ -454,8 +493,6 @@ following image formats are supported:
@item 4X Movie               @tab     @tab  X
    @tab Used in certain computer games.
@item 8088flex TMV           @tab     @tab  X
-@item 8SVX exponential       @tab     @tab  X
-@item 8SVX fibonacci         @tab     @tab  X
@item A64 multicolor         @tab  X  @tab
    @tab Creates video suitable to be played on a commodore 64 (multicolor mode).
@item Amazing Studio PAF Video @tab     @tab  X
@@ -464,6 +501,7 @@ following image formats are supported:
@item AMV Video              @tab  X  @tab  X
    @tab Used in Chinese MP3 players.
@item ANSI/ASCII art         @tab     @tab  X
+@item Apple Intermediate Codec @tab     @tab  X
@item Apple MJPEG-B          @tab     @tab  X
@item Apple ProRes           @tab  X  @tab  X
@item Apple QuickDraw        @tab     @tab  X
@@ -546,6 +584,8 @@ following image formats are supported:
    @tab Sorenson H.263 used in Flash
@item Forward Uncompressed   @tab     @tab  X
@item Fraps                  @tab     @tab  X
+@item Go2Webinar             @tab     @tab  X
+    @tab fourcc: G2M4
@item H.261                  @tab  X  @tab  X
@item H.263 / H.263-1996     @tab  X  @tab  X
@item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
@@ -622,6 +662,8 @@ following image formats are supported:
    @tab fourcc: VP60,VP61,VP62
@item VP8                    @tab  E  @tab  X
    @tab fourcc: VP80, encoding supported through external library libvpx
+@item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
+    @tab fourcc: Y216
@item Prores                 @tab     @tab  X
    @tab fourcc: apch,apcn,apcs,apco
@item Q-team QPEG            @tab     @tab  X
@@ -645,8 +687,11 @@ following image formats are supported:
    @tab Texture dictionaries used by the Renderware Engine.
@item RL2 video              @tab     @tab  X
    @tab used in some games by Entertainment Software Partners
+@item SGI RLE 8-bit          @tab     @tab  X
@item Sierra VMD video       @tab     @tab  X
    @tab Used in Sierra VMD files.
+@item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
+@item Silicon Graphics Motion Video Compressor 2 (MVC2)  @tab     @tab  X
@item Smacker video          @tab     @tab  X
    @tab Video encoding used in Smacker.
@item SMPTE VC-1             @tab     @tab  X
@@ -706,7 +751,8 @@ following image formats are supported:

@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
-@item 8SVX audio             @tab     @tab  X
+@item 8SVX exponential       @tab     @tab  X
+@item 8SVX fibonacci         @tab     @tab  X
@item AAC+                   @tab  E  @tab  X
    @tab encoding supported through external library libaacplus
@item AAC                    @tab  E  @tab  X
@@ -737,19 +783,21 @@ following image formats are supported:
@item ADPCM IMA Westwood     @tab     @tab  X
@item ADPCM ISS IMA          @tab     @tab  X
    @tab Used in FunCom games.
+@item ADPCM IMA Dialogic     @tab     @tab  X
@item ADPCM IMA Duck DK3     @tab     @tab  X
    @tab Used in some Sega Saturn console games.
@item ADPCM IMA Duck DK4     @tab     @tab  X
    @tab Used in some Sega Saturn console games.
+@item ADPCM IMA Radical      @tab     @tab  X
@item ADPCM Microsoft        @tab  X  @tab  X
@item ADPCM MS IMA           @tab  X  @tab  X
+@item ADPCM Nintendo Gamecube AFC  @tab     @tab  X
+@item ADPCM Nintendo Gamecube DTK  @tab     @tab  X
@item ADPCM Nintendo Gamecube THP  @tab     @tab  X
@item ADPCM QT IMA           @tab  X  @tab  X
@item ADPCM SEGA CRI ADX     @tab  X  @tab  X
    @tab Used in Sega Dreamcast games.
@item ADPCM Shockwave Flash  @tab  X  @tab  X
-@item ADPCM SMJPEG IMA       @tab     @tab  X
-    @tab Used in certain Loki game ports.
@item ADPCM Sound Blaster Pro 2-bit  @tab     @tab  X
@item ADPCM Sound Blaster Pro 2.6-bit  @tab     @tab  X
@item ADPCM Sound Blaster Pro 4-bit  @tab     @tab  X
@@ -787,6 +835,7 @@ following image formats are supported:
@item DSP Group TrueSpeech   @tab     @tab  X
@item DV audio               @tab     @tab  X
@item Enhanced AC-3          @tab  X  @tab  X
+@item EVRC (Enhanced Variable Rate Codec) @tab     @tab  X
@item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
@item G.723.1                @tab X @tab X
@item G.729                  @tab     @tab  X
@@ -803,7 +852,6 @@ following image formats are supported:
@item MLP (Meridian Lossless Packing)  @tab     @tab  X
    @tab Used in DVD-Audio discs.
@item Monkey's Audio         @tab     @tab  X
-    @tab Only versions 3.97-3.99 are supported.
@item MP1 (MPEG audio layer 1)  @tab     @tab IX
@item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
    @tab libtwolame can be used alternatively for encoding.
@@ -813,9 +861,15 @@ following image formats are supported:
@item Musepack SV7           @tab     @tab  X
@item Musepack SV8           @tab     @tab  X
@item Nellymoser Asao        @tab  X  @tab  X
+@item Opus                   @tab  E  @tab  E
+    @tab supported through external library libopus
@item PCM A-law              @tab  X  @tab  X
@item PCM mu-law             @tab  X  @tab  X
-@item PCM 16-bit little-endian planar  @tab     @tab  X
+@item PCM signed 8-bit planar  @tab  X  @tab  X
+@item PCM signed 16-bit big-endian planar  @tab  X  @tab  X
+@item PCM signed 16-bit little-endian planar  @tab  X  @tab  X
+@item PCM signed 24-bit little-endian planar  @tab  X  @tab  X
+@item PCM signed 32-bit little-endian planar  @tab  X  @tab  X
@item PCM 32-bit floating point big-endian  @tab  X  @tab  X
@item PCM 32-bit floating point little-endian  @tab  X  @tab  X
@item PCM 64-bit floating point big-endian  @tab  X  @tab  X
@@ -852,14 +906,15 @@ following image formats are supported:
@item Sierra VMD audio       @tab     @tab  X
    @tab Used in Sierra VMD files.
@item Smacker audio          @tab     @tab  X
-@item SMPTE 302M AES3 audio  @tab     @tab  X
+@item SMPTE 302M AES3 audio  @tab  X  @tab  X
@item Sonic                  @tab  X  @tab  X
    @tab experimental codec
@item Sonic lossless         @tab  X  @tab  X
    @tab experimental codec
@item Speex                  @tab  E  @tab  E
    @tab supported through external library libspeex
-@item True Audio (TTA)       @tab     @tab  X
+@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
+@item True Audio (TTA)       @tab  X  @tab  X
@item TrueHD                 @tab     @tab  X
    @tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor)    @tab     @tab  X
@@ -867,7 +922,8 @@ following image formats are supported:
    @tab Used in LucasArts SMUSH animations.
@item Vorbis                 @tab  E  @tab  X
    @tab A native but very primitive encoder exists.
-@item WavPack                @tab     @tab  X
+@item WavPack                @tab  E  @tab  X
+    @tab supported through external library libwavpack
@item Westwood Audio (SND1)  @tab     @tab  X
@item Windows Media Audio 1  @tab  X  @tab  X
@item Windows Media Audio 2  @tab  X  @tab  X
@@ -887,18 +943,26 @@ performance on systems without hardware floating point support).

@multitable @columnfractions .4 .1 .1 .1 .1
@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
-@item SSA/ASS          @tab X @tab X @tab X @tab X
+@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
+@item AQTitle          @tab   @tab X @tab   @tab X
@item DVB              @tab X @tab X @tab X @tab X
@item DVD              @tab X @tab X @tab X @tab X
@item JACOsub          @tab X @tab X @tab   @tab X
@item MicroDVD         @tab X @tab X @tab   @tab X
+@item MPL2             @tab   @tab X @tab   @tab X
+@item MPsub (MPlayer)  @tab   @tab X @tab   @tab X
@item PGS              @tab   @tab   @tab   @tab X
+@item PJS (Phoenix)    @tab   @tab X @tab   @tab X
@item RealText         @tab   @tab X @tab   @tab X
@item SAMI             @tab   @tab X @tab   @tab X
+@item SSA/ASS          @tab X @tab X @tab X @tab X
@item SubRip (SRT)     @tab X @tab X @tab X @tab X
+@item SubViewer v1     @tab   @tab X @tab   @tab X
@item SubViewer        @tab   @tab X @tab   @tab X
-@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
-@item WebVTT           @tab   @tab X @tab   @tab X
+@item TED Talks captions @tab @tab X @tab   @tab X
+@item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
+@item VPlayer          @tab   @tab X @tab   @tab X
+@item WebVTT           @tab X @tab X @tab   @tab X
@item XSUB             @tab   @tab   @tab X @tab X
@end multitable

@@ -951,7 +1015,7 @@ performance on systems without hardware floating point support).
@item OSS               @tab X      @tab X
@item Pulseaudio        @tab X      @tab
@item SDL               @tab        @tab X
-@item Video4Linux2      @tab X      @tab
+@item Video4Linux2      @tab X      @tab X
@item VfW capture       @tab X      @tab
@item X11 grabbing      @tab X      @tab
@end multitable
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -86,7 +86,7 @@ fail to open.
 Set the video size in the captured video.

@item framerate
-Set the framerate in the captured video.
+Set the frame rate in the captured video.

@item sample_rate
 Set the sample rate (in Hz) of the captured audio.
@@ -224,6 +224,13 @@ 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
@@ -320,6 +327,12 @@ 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
@@ -328,14 +341,14 @@ device.
@item
 Create a color video stream and play it back with @command{ffplay}:
@example
-ffplay -f lavfi -graph "color=pink [out0]" dummy
+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=pink
+ffplay -f lavfi color=c=pink
@end example

@item
@@ -570,10 +583,16 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example

-@section video4linux2
+@section video4linux2, v4l2

 Video4Linux2 input video device.

+"v4l2" can be used as alias for "video4linux2".
+
+If FFmpeg is built with v4l-utils support (by using the
+@code{--enable-libv4l2} configure option), it is possible to use it with the
+@code{-use_libv4l2} input device option.
+
 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
@@ -581,10 +600,10 @@ 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
+@var{width}x@var{height} sizes and frame rates. 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:
+Some devices, like TV cards, support one or more standards. It is possible
+to list all the supported standards using @command{-list_standards all}.

 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
@@ -593,19 +612,94 @@ 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.
+Some usage examples of the video4linux2 device with @command{ffmpeg}
+and @command{ffplay}:
+@itemize
+@item
+Grab and show the input of a video4linux2 device:
@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".
+@item
+Grab and record the input of a video4linux2 device, leave the
+frame rate and size as previously set:
+@example
+ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
+@end example
+@end itemize
+
+For more information about Video4Linux, check @url{http://linuxtv.org/}.
+
+@subsection Options
+
+@table @option
+@item standard
+Set the standard. Must be the name of a supported standard. To get a
+list of the supported standards, use the @option{list_standards}
+option.
+
+@item channel
+Set the input channel number. Default to -1, which means using the
+previously selected channel.
+
+@item video_size
+Set the video frame size. The argument must be a string in the form
+@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation.
+
+@item pixel_format
+Select the pixel format (only valid for raw video input).
+
+@item input_format
+Set the preferred pixel format (for raw video) or a codec name.
+This option allows to select the input format, when several are
+available.
+
+@item framerate
+Set the preferred video frame rate.
+
+@item list_formats
+List available formats (supported pixel formats, codecs, and frame
+sizes) and exit.
+
+Available values are:
+@table @samp
+@item all
+Show all available (compressed and non-compressed) formats.
+
+@item raw
+Show only raw video (non-compressed) formats.
+
+@item compressed
+Show only compressed formats.
+@end table
+
+@item list_standards
+List supported standards and exit.
+
+Available values are:
+@table @samp
+@item all
+Show all supported standards.
+@end table
+
+@item timestamps, ts
+Set type of timestamps for grabbed frames.
+
+Available values are:
+@table @samp
+@item default
+Use timestamps from the kernel.
+
+@item abs
+Use absolute timestamps (wall clock).
+
+@item mono2abs
+Force conversion from monotonic to absolute timestamps.
+@end table
+
+Default value is @code{default}.
+@end table

@section vfwcap

@@ -678,7 +772,7 @@ ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg

@item framerate
 Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a framerate of @code{30000/1001}.
+corresponding to a frame rate of @code{30000/1001}.

@item show_region
 Show grabbed region on screen.
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -24,7 +24,7 @@ a mail for every change to every issue.
 The subscription URL for the ffmpeg-trac list is:
 http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-http(s)://ffmpeg.org/trac/ffmpeg
+http(s)://trac.ffmpeg.org

 Type:
 -----
--- a/doc/libavcodec.texi
+++ b/doc/libavcodec.texi
@@ -0,0 +1,48 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libavcodec Documentation
+@titlepage
+@center @titlefont{Libavcodec Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+The libavcodec library provides a generic encoding/decoding framework
+and contains multiple decoders and encoders for audio, video and
+subtitle streams, and several bitstream filters.
+
+The shared architecture provides various services ranging from bit
+stream I/O to DSP optimizations, and makes it suitable for
+implementing robust and fast codecs as well as for experimentation.
+
+@c man end DESCRIPTION
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
+@url{libavutil.html,libavutil}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
+ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
+libavutil(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename libavcodec
+@settitle media streams decoding and encoding library
+
+@end ignore
+
+@bye
--- a/doc/libavdevice.texi
+++ b/doc/libavdevice.texi
@@ -0,0 +1,45 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libavdevice Documentation
+@titlepage
+@center @titlefont{Libavdevice Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+The libavdevice library provides a generic framework for grabbing from
+and rendering to many common multimedia input/output devices, and
+supports several input and output devices, including Video4Linux2,
+VfW, DShow, and ALSA.
+
+@c man end DESCRIPTION
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{ffmpeg-devices.html,ffmpeg-devices},
+@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
+ffmpeg-devices(1),
+libavutil(3), libavcodec(3), libavformat(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename libavdevice
+@settitle multimedia device handling library
+
+@end ignore
+
+@bye
--- a/doc/libavfilter.texi
+++ b/doc/libavfilter.texi
@@ -0,0 +1,44 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libavfilter Documentation
+@titlepage
+@center @titlefont{Libavfilter Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+The libavfilter library provides a generic audio/video filtering
+framework containing several filters, sources and sinks.
+
+@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-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
+
+@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
+
+@include authors.texi
+
+@ignore
+
+@setfilename libavfilter
+@settitle multimedia filtering library
+
+@end ignore
+
+@bye
--- a/doc/libavformat.texi
+++ b/doc/libavformat.texi
@@ -0,0 +1,48 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libavformat Documentation
+@titlepage
+@center @titlefont{Libavformat Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+The libavformat library provides a generic framework for multiplexing
+and demultiplexing (muxing and demuxing) audio, video and subtitle
+streams. It encompasses multiple muxers and demuxers for multimedia
+container formats.
+
+It also supports several input and output protocols to access a media
+resource.
+
+@c man end DESCRIPTION
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
+@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
+ffmpeg-formats(1), ffmpeg-protocols(1),
+libavutil(3), libavcodec(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename libavformat
+@settitle multimedia muxing and demuxing library
+
+@end ignore
+
+@bye
--- a/doc/libavutil.texi
+++ b/doc/libavutil.texi
@@ -0,0 +1,44 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libavutil Documentation
+@titlepage
+@center @titlefont{Libavutil Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+The libavutil library is a utility library to aid portable
+multimedia programming. It contains safe portable string functions,
+random number generators, data structures, additional mathematics
+functions, cryptography and multimedia related functionality (like
+enumerations for pixel and sample formats).
+
+@c man end DESCRIPTION
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{ffmpeg-utils.html,ffmpeg-utils}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
+ffmpeg-utils(1)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename libavutil
+@settitle multimedia-biased utility library
+
+@end ignore
+
+@bye
--- a/doc/libswresample.texi
+++ b/doc/libswresample.texi
@@ -0,0 +1,70 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libswresample Documentation
+@titlepage
+@center @titlefont{Libswresample Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+The libswresample library performs highly optimized audio resampling,
+rematrixing and sample format conversion operations.
+
+Specifically, this library performs the following conversions:
+
+@itemize
+@item
+@emph{Resampling}: is the process of changing the audio rate, for
+example from an high sample rate of 44100Hz to 8000Hz. Audio
+conversion from high to low sample rate is a lossy process. Several
+resampling options and algorithms are available.
+
+@item
+@emph{Format conversion}: is the process of converting the type of
+samples, for example from 16-bit signed samples to unsigned 8-bit or
+float samples. It also handles packing conversion, when passing from
+packed layout (all samples belonging to distinct channels interleaved
+in the same buffer), to planar layout (all samples belonging to the
+same channel stored in a dedicated buffer or "plane").
+
+@item
+@emph{Rematrixing}: is the process of changing the channel layout, for
+example from stereo to mono. When the input channels cannot be mapped
+to the output streams, the process is lossy, since it involves
+different gain factors and mixing.
+@end itemize
+
+Various other audio conversions (e.g. stretching and padding) are
+enabled through dedicated options.
+
+@c man end DESCRIPTION
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{ffmpeg-resampler.html,ffmpeg-resampler},
+@url{libavutil.html,libavutil}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
+ffmpeg-resampler(1),
+libavutil(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename libswresample
+@settitle audio resampling library
+
+@end ignore
+
+@bye
--- a/doc/libswscale.texi
+++ b/doc/libswscale.texi
@@ -0,0 +1,63 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libswscale Documentation
+@titlepage
+@center @titlefont{Libswscale Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+@c man begin DESCRIPTION
+
+The libswscale library performs highly optimized image scaling and
+colorspace and pixel format conversion operations.
+
+Specifically, this library performs the following conversions:
+
+@itemize
+@item
+@emph{Rescaling}: is the process of changing the video size. Several
+rescaling options and algorithms are available. This is usually a
+lossy process.
+
+@item
+@emph{Pixel format conversion}: is the process of converting the image
+format and colorspace of the image, for example from planar YUV420P to
+RGB24 packed. It also handles packing conversion, that is converts
+from packed layout (all pixels belonging to distinct planes
+interleaved in the same buffer), to planar layout (all samples
+belonging to the same plane stored in a dedicated buffer or "plane").
+
+This is usually a lossy process in case the source and destination
+colorspaces differ.
+@end itemize
+
+@c man end DESCRIPTION
+
+@chapter See Also
+
+@ifhtml
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@url{ffmpeg-scaler.html,ffmpeg-scaler},
+@url{libavutil.html,libavutil}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
+ffmpeg-scaler(1),
+libavutil(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename libswscale
+@settitle video scaling and pixel format conversion library
+
+@end ignore
+
+@bye
--- a/doc/metadata.texi
+++ b/doc/metadata.texi
@@ -65,4 +65,20 @@ title=chapter \#1
 title=multi\
 line
@end example
+
+By using the ffmetadata muxer and demuxer it is possible to extract
+metadata from an input file to an ffmetadata file, and then transcode
+the file into an output file with the edited ffmetadata file.
+
+Extracting an ffmetadata file with @file{ffmpeg} goes as follows:
+@example
+ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE
+@end example
+
+Reinserting edited metadata information from the FFMETADATAFILE file can
+be done as:
+@example
+ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT
+@end example
+
@c man end METADATA
--- a/doc/mips.txt
+++ b/doc/mips.txt
@@ -45,8 +45,12 @@ Example of copyright notice:
 Files that have MIPS copyright notice in them:
 ===============================================
 * libavutil/mips/
+      float_dsp_mips.c
      libm_mips.h
 * libavcodec/mips/
+      aaccoder_mips.c
+      aacpsy_mips.h
+      ac3dsp_mips.c
      acelp_filters_mips.c
      acelp_vectors_mips.c
      amrwbdec_mips.c
@@ -61,5 +65,6 @@ Files that have MIPS copyright notice in them:
      fft_table.h
      fft_init_table.c
      fmtconvert_mips.c
+      iirfilter_mips.c
      mpegaudiodsp_mips_fixed.c
      mpegaudiodsp_mips_float.c
--- a/doc/multithreading.txt
+++ b/doc/multithreading.txt
@@ -57,6 +57,11 @@ 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.

+If there are inter-frame dependencies, so the codec calls
+ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
+frames must then be freed with ff_thread_release_buffer().
+Otherwise leave it at zero and decode directly into the user-supplied frames.
+
 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
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -18,6 +18,23 @@ 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

@@ -129,6 +146,32 @@ ffmpeg -i INPUT -f framemd5 -

 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

@@ -214,12 +257,108 @@ Note also that the pattern must not necessarily contain "%d" or
 ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example

+@table @option
+@item start_number @var{number}
+Start the sequence from @var{number}. Default value is 1. Must be a
+positive number.
+
+@item -update @var{number}
+If @var{number} is nonzero, the filename will always be interpreted as just a
+filename, not a pattern, and this file will be continuously overwritten with new
+images.
+
+@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.

+@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
+
+This muxer supports the following options:
+
+@table @option
+
+@item reserve_index_space
+By default, this muxer writes the index for seeking (called cues in Matroska
+terms) at the end of the file, because it cannot know in advance how much space
+to leave for the index at the beginning of the file. However for some use cases
+-- e.g.  streaming where seeking is possible but slow -- it is useful to put the
+index at the beginning of the file.
+
+If this option is set to a non-zero value, the muxer will reserve a given amount
+of space in the file header and then try to write the cues there when the muxing
+finishes. If the available space does not suffice, muxing will fail. A safe size
+for most use cases should be about 50kB per hour of video.
+
+Note that cues are only written if the output is seekable and this option will
+have no effect if it is not.
+
+@end table
+
@anchor{md5}
@section md5

@@ -252,7 +391,8 @@ See also the @ref{framemd5} muxer.
 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 using the @command{qt-faststart} tool). A fragmented
+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
@@ -310,6 +450,12 @@ 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.
+@item -movflags rtphint
+Add RTP hinting tracks to the output file.
@end table

 Smooth Streaming content can be pushed in real time to a publishing
@@ -318,6 +464,42 @@ point on IIS with this muxer. Example:
 ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example

+@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
+
@section mpegts

 MPEG transport stream muxer.
@@ -381,69 +563,21 @@ Alternatively you can write the command as:
 ffmpeg -benchmark -i INPUT -f null -
@end example

-@section matroska
+@section ogg

-Matroska container muxer.
-
-This muxer implements the matroska and webm container specs.
-
-The recognized metadata settings in this muxer are:
+Ogg container muxer.

@table @option
-
-@item title=@var{title name}
-Name provided to a single track
+@item -page_duration @var{duration}
+Preferred page duration, in microseconds. The muxer will attempt to create
+pages that are approximately @var{duration} microseconds long. This allows the
+user to compromise between seek granularity and container overhead. The default
+is 1 second. A value of 0 will fill all segments, making pages as large as
+possible. A value of 1 will effectively use 1 packet-per-page in most
+situations, giving a small seek granularity at the cost of additional container
+overhead.
@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.
@@ -457,7 +591,9 @@ 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 video keyframe, if a video stream is present.
+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
@@ -473,38 +609,45 @@ the option @var{segment_list}. The list type is specified by the
 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 @code{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
+@table @samp
@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}.
+Default value is @code{samp}.

@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}
+Update the list file so that it contains at most the last @var{size}
+segments. If 0 the list file will contain all the segments. 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
+@table @samp
@item flat
 Generate a flat list for the created segments, one segment per line.

@@ -525,21 +668,36 @@ 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}.
+@samp{ext} is deprecated in favor or @samp{csv}.
+
+@item ffconcat
+Generate an ffconcat file for the created segments. The resulting file
+can be read using the FFmpeg @ref{concat} demuxer.
+
+A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will
+auto-select this format.

@item m3u8
-Generate an extended M3U8 file, version 4, compliant with
-@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming-08.txt}.
+Generate an extended M3U8 file, version 3, compliant with
+@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}.

 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".
+Set segment duration to @var{time}, the value must be a duration
+specification. Default value is "2". See also the
+@option{segment_times} option.
+
+Note that splitting may not be accurate, unless you force the
+reference stream key-frames at the given time. See the introductory
+notice and the examples below.
+
@item segment_time_delta @var{delta}
 Specify the accuracy time when selecting the start time for a
-segment. Default value is "0".
+segment, expressed as a duration specification. Default value is "0".

 When delta is specified a key-frame will start a new segment if its
 PTS satisfies the relation:
@@ -561,12 +719,31 @@ 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.
+separated duration specifications, in increasing order. See also
+the @option{segment_time} option.
+
+@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

-Some examples follow.
+@subsection Examples

@itemize
@item
@@ -585,17 +762,24 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_ti
@end example

@item
-As the example above, but use the @code{ffmpeg} @var{force_key_frames}
+As the example above, but use the @command{ffmpeg} @option{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
+with the segment option @option{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 -vcodec mpeg4 -acodec pcm_s16le -map 0 \
+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 @option{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:
@@ -612,39 +796,39 @@ ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
@end example
@end itemize

-@section mp3
+@section tee

-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.
+The tee muxer can be used to write the same data to several files or any
+other kind of muxer. It can be used, for example, to both stream a video to
+the network and save it to disk at the same time.

-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.
+It is different from specifying several outputs to the @command{ffmpeg}
+command-line tool because the audio and video data will be encoded only once
+with the tee muxer; encoding can be a very expensive process. It is not
+useful when using the libavformat API directly because it is then possible
+to feed the same packets to several muxers directly.

-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.
+The slave outputs are specified in the file name given to the muxer,
+separated by '|'. If any of the slave name contains the '|' separator,
+leading or trailing spaces or any special character, it must be
+escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
+manual).

-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.
+Options can be specified for each slave by prepending them as a list of
+@var{key}=@var{value} pairs separated by ':', between square brackets. If
+the options values contain a special character or the ':' separator, they
+must be escaped; note that this is a second level escaping.

-Examples:
+Example: encode something and both archive it in a WebM file and stream it
+as MPEG-TS over UDP (the streams need to be explicitly mapped):

-Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
@example
-ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
+ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
+  "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@end example

-Attach a picture to an mp3:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover"
-metadata:s:v comment="Cover (Front)" out.mp3
-@end example
+Note: some codecs may need different options depending on the output format;
+the auto-detection of this can not work with the tee muxer. The main example
+is the @option{global_header} flag.

@c man end MUXERS
--- a/doc/nut.texi
+++ b/doc/nut.texi
@@ -0,0 +1,138 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle NUT
+
+@titlepage
+@center @titlefont{NUT}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Description
+NUT is a low overhead generic container format. It stores audio, video,
+subtitle and user-defined streams in a simple, yet efficient, way.
+
+It was created by a group of FFmpeg and MPlayer developers in 2003
+and was finalized in 2008.
+
+The official nut specification is at svn://svn.mplayerhq.hu/nut
+In case of any differences between this text and the official specification,
+the official specification shall prevail.
+
+@chapter Container-specific codec tags
+
+@section Generic raw YUVA formats
+
+Since many exotic planar YUVA pixel formats are not considered by
+the AVI/QuickTime FourCC lists, the following scheme is adopted for
+representing them.
+
+The first two bytes can contain the values:
+Y1 = only Y
+Y2 = Y+A
+Y3 = YUV
+Y4 = YUVA
+
+The third byte represents the width and height chroma subsampling
+values for the UV planes, that is the amount to shift the luma
+width/height right to find the chroma width/height.
+
+The fourth byte is the number of bits used (8, 16, ...).
+
+If the order of bytes is inverted, that means that each component has
+to be read big-endian.
+
+@section Raw Audio
+
+@multitable @columnfractions .4 .4
+@item ALAW  @tab A-LAW
+@item ULAW  @tab MU-LAW
+@item P<type><interleaving><bits> @tab little-endian PCM
+@item <bits><interleaving><type>P @tab big-endian PCM
+@end multitable
+
+<type> is S for signed integer, U for unsigned integer, F for IEEE float
+<interleaving> is D for default, P is for planar.
+<bits> is 8/16/24/32
+
+@example
+PFD[32]   would for example be signed 32 bit little-endian IEEE float
+@end example
+
+@section Subtitles
+
+@multitable @columnfractions .4 .4
+@item UTF8   @tab Raw UTF-8
+@item SSA[0] @tab SubStation Alpha
+@item DVDS   @tab DVD subtitles
+@item DVBS   @tab DVB subtitles
+@end multitable
+
+@section Raw Data
+
+@multitable @columnfractions .4 .4
+@item UTF8   @tab Raw UTF-8
+@end multitable
+
+@section Codecs
+
+@multitable @columnfractions .4 .4
+@item 3IV1 @tab non-compliant MPEG-4 generated by old 3ivx
+@item ASV1 @tab Asus Video
+@item ASV2 @tab Asus Video 2
+@item CVID @tab Cinepak
+@item CYUV @tab Creative YUV
+@item DIVX @tab non-compliant MPEG-4 generated by old DivX
+@item DUCK @tab Truemotion 1
+@item FFV1 @tab FFmpeg video 1
+@item FFVH @tab FFmpeg Huffyuv
+@item H261 @tab ITU H.261
+@item H262 @tab ITU H.262
+@item H263 @tab ITU H.263
+@item H264 @tab ITU H.264
+@item HFYU @tab Huffyuv
+@item I263 @tab Intel H.263
+@item IV31 @tab Indeo 3.1
+@item IV32 @tab Indeo 3.2
+@item IV50 @tab Indeo 5.0
+@item LJPG @tab ITU JPEG (lossless)
+@item MJLS @tab ITU JPEG-LS
+@item MJPG @tab ITU JPEG
+@item MPG4 @tab MS MPEG-4v1 (not ISO MPEG-4)
+@item MP42 @tab MS MPEG-4v2
+@item MP43 @tab MS MPEG-4v3
+@item MP4V @tab ISO MPEG-4 Part 2 Video (from old encoders)
+@item mpg1 @tab ISO MPEG-1 Video
+@item mpg2 @tab ISO MPEG-2 Video
+@item MRLE @tab MS RLE
+@item MSVC @tab MS Video 1
+@item RT21 @tab Indeo 2.1
+@item RV10 @tab RealVideo 1.0
+@item RV20 @tab RealVideo 2.0
+@item RV30 @tab RealVideo 3.0
+@item RV40 @tab RealVideo 4.0
+@item SNOW @tab FFmpeg Snow
+@item SVQ1 @tab Sorenson Video 1
+@item SVQ3 @tab Sorenson Video 3
+@item theo @tab Xiph Theora
+@item TM20 @tab Truemotion 2.0
+@item UMP4 @tab non-compliant MPEG-4 generated by UB Video MPEG-4
+@item VCR1 @tab ATI VCR1
+@item VP30 @tab VP 3.0
+@item VP31 @tab VP 3.1
+@item VP50 @tab VP 5.0
+@item VP60 @tab VP 6.0
+@item VP61 @tab VP 6.1
+@item VP62 @tab VP 6.2
+@item VP70 @tab VP 7.0
+@item WMV1 @tab MS WMV7
+@item WMV2 @tab MS WMV8
+@item WMV3 @tab MS WMV9
+@item WV1F @tab non-compliant MPEG-4 generated by ?
+@item WVC1 @tab VC-1
+@item XVID @tab non-compliant MPEG-4 generated by old Xvid
+@item XVIX @tab non-compliant MPEG-4 generated by old Xvid with interlacing bug
+@end multitable
+
--- a/doc/optimization.txt
+++ b/doc/optimization.txt
@@ -148,7 +148,7 @@ Alignment:
 Some instructions on some architectures have strict alignment restrictions,
 for example most SSE/SSE2 instructions on x86.
 The minimum guaranteed alignment is written in the .h files, for example:
-    void (*put_pixels_clamped)(const DCTELEM *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
+    void (*put_pixels_clamped)(const int16_t *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);


 General Tips:
@@ -253,7 +253,7 @@ Optimization guide for ARM11 (used in Nokia N800 Internet Tablet):
 http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf
 Optimization guide for Intel XScale (used in Sharp Zaurus PDA):
 http://download.intel.com/design/intelxscale/27347302.pdf
-Intel Wireless MMX2 Coprocessor: Programmers Reference Manual
+Intel Wireless MMX 2 Coprocessor: Programmers Reference Manual
 http://download.intel.com/design/intelxscale/31451001.pdf

 PowerPC-specific:
--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -26,7 +26,7 @@ ALSA (Advanced Linux Sound Architecture) output device.

 CACA output device.

-This output devices allows to show a video stream in CACA window.
+This output device 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.

@@ -112,7 +112,7 @@ OSS (Open Sound System) output device.

 SDL (Simple DirectMedia Layer) output device.

-This output devices allows to show a video stream in an SDL
+This output device 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.

@@ -153,4 +153,69 @@ ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL

 sndio audio output device.

+@section xv
+
+XV (XVideo) output device.
+
+This output device allows to show a video stream in a X Window System
+window.
+
+@subsection Options
+
+@table @option
+@item display_name
+Specify the hardware display name, which determines the display and
+communications domain to be used.
+
+The display name or DISPLAY environment variable can be a string in
+the format @var{hostname}[:@var{number}[.@var{screen_number}]].
+
+@var{hostname} specifies the name of the host machine on which the
+display is physically attached. @var{number} specifies the number of
+the display server on that host machine. @var{screen_number} specifies
+the screen to be used on that server.
+
+If unspecified, it defaults to the value of the DISPLAY environment
+variable.
+
+For example, @code{dual-headed:0.1} would specify screen 1 of display
+0 on the machine named ``dual-headed''.
+
+Check the X11 specification for more detailed information about the
+display name format.
+
+@item window_size
+Set the created 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 window_x
+@item window_y
+Set the X and Y window offsets for the created window. They are both
+set to 0 by default. The values may be ignored by the window manager.
+
+@item window_title
+Set the window title, if not specified default to the filename
+specified for the output device.
+@end table
+
+For more information about XVideo see @url{http://www.x.org/}.
+
+@subsection Examples
+
+@itemize
+@item
+Decode, display and encode video input with @command{ffmpeg} at the
+same time:
+@example
+ffmpeg -i INPUT OUTPUT -f xv display
+@end example
+
+@item
+Decode and display the input video to multiple X11 windows:
+@example
+ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
+@end example
+@end itemize
+
@c man end OUTPUT DEVICES
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-

-@settitle Platform Specific information
+@settitle Platform Specific Information
@titlepage
-@center @titlefont{Platform Specific information}
+@center @titlefont{Platform Specific Information}
@end titlepage

@top
@@ -77,30 +77,15 @@ For information about compiling FFmpeg on OS/2 see
@chapter Windows

 To get help and instructions for building FFmpeg under Windows, check out
-the FFmpeg Windows Help Forum at
-@url{http://ffmpeg.arrozcru.org/}.
+the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.

-@section Native Windows compilation
+@section Native Windows compilation using MinGW or MinGW-w64

-FFmpeg can be built to run natively on Windows using the MinGW tools. Install
-the latest versions of MSYS and MinGW from @url{http://www.mingw.org/}.
-You can find detailed installation instructions in the download
-section and the FAQ.
-
-FFmpeg does not build out-of-the-box with the packages the automated MinGW
-installer provides. It also requires coreutils to be installed and many other
-packages updated to the latest version. The minimum versions for some packages
-are listed below:
-
-@itemize
-@item bash 3.1
-@item msys-make 3.81-2 (note: not mingw32-make)
-@item w32api 3.13
-@item mingw-runtime 3.15
-@end itemize
-
-FFmpeg automatically passes @code{-fno-common} to the compiler to work around
-a GCC bug (see @url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=37216}).
+FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64
+toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from
+@url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}.
+You can find detailed installation instructions in the download section and
+the FAQ.

 Notes:

@@ -121,149 +106,106 @@ libavformat) as DLLs.

@end itemize

-@section Microsoft Visual C++ compatibility
+@section Microsoft Visual C++ or Intel C++ Compiler for Windows

-As stated in the FAQ, FFmpeg will not compile under MSVC++. However, if you
-want to use the libav* libraries in your own applications, you can still
-compile those applications using MSVC++. But the libav* libraries you link
-to @emph{must} be built with MinGW. However, you will not be able to debug
-inside the libav* libraries, since MSVC++ does not recognize the debug
-symbols generated by GCC.
-We strongly recommend you to move over from MSVC++ to MinGW tools.
+FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
+wrapper. For ICL, only the wrapper is used, since ICL supports C99.

-This description of how to use the FFmpeg libraries with MSVC++ is based on
-Microsoft Visual C++ 2005 Express Edition. If you have a different version,
-you might have to modify the procedures slightly.
+You will need the following prerequisites:

-@subsection Using static libraries
+@itemize
+@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
+@item @uref{http://www.mingw.org/, MSYS}
+@item @uref{http://yasm.tortall.net/, YASM}
+@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
+you want to run @uref{fate.html, FATE}.
+@end itemize

-Assuming you have just built and installed FFmpeg in @file{/usr/local}:
+To set up a proper environment in MSYS, you need to run @code{msys.bat} from
+the Visual Studio or Intel Compiler command prompt.

-@enumerate
+Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
+somewhere in your @code{PATH}.

-@item Create a new console application ("File / New / Project") and then
-select "Win32 Console Application". On the appropriate page of the
-Application Wizard, uncheck the "Precompiled headers" option.
+Next, make sure @code{inttypes.h} and any other headers and libs you want to use
+are located in a spot that the compiler can see. Do so by modifying the @code{LIB}
+and @code{INCLUDE} environment variables to include the @strong{Windows} paths to
+these directories. Alternatively, you can try and use the
+@code{--extra-cflags}/@code{--extra-ldflags} configure options.

-@item Write the source code for your application, or, for testing, just
-copy the code from an existing sample application into the source file
-that MSVC++ has already created for you. For example, you can copy
-@file{libavformat/output-example.c} from the FFmpeg distribution.
-
-@item Open the "Project / Properties" dialog box. In the "Configuration"
-combo box, select "All Configurations" so that the changes you make will
-affect both debug and release builds. In the tree view on the left hand
-side, select "C/C++ / General", then edit the "Additional Include
-Directories" setting to contain the path where the FFmpeg includes were
-installed (i.e. @file{c:\msys\1.0\local\include}).
-Do not add MinGW's include directory here, or the include files will
-conflict with MSVC's.
-
-@item Still in the "Project / Properties" dialog box, select
-"Linker / General" from the tree view and edit the
-"Additional Library Directories" setting to contain the @file{lib}
-directory where FFmpeg was installed (i.e. @file{c:\msys\1.0\local\lib}),
-the directory where MinGW libs are installed (i.e. @file{c:\mingw\lib}),
-and the directory where MinGW's GCC libs are installed
-(i.e. @file{C:\mingw\lib\gcc\mingw32\4.2.1-sjlj}). Then select
-"Linker / Input" from the tree view, and add the files @file{libavformat.a},
-@file{libavcodec.a}, @file{libavutil.a}, @file{libmingwex.a},
-@file{libgcc.a}, and any other libraries you used (i.e. @file{libz.a})
-to the end of "Additional Dependencies".
-
-@item Now, select "C/C++ / Code Generation" from the tree view. Select
-"Debug" in the "Configuration" combo box. Make sure that "Runtime
-Library" is set to "Multi-threaded Debug DLL". Then, select "Release" in
-the "Configuration" combo box and make sure that "Runtime Library" is
-set to "Multi-threaded DLL".
-
-@item Click "OK" to close the "Project / Properties" dialog box.
-
-@item MSVC++ lacks some C99 header files that are fundamental for FFmpeg.
-Get msinttypes from @url{http://code.google.com/p/msinttypes/downloads/list}
-and install it in MSVC++'s include directory
-(i.e. @file{C:\Program Files\Microsoft Visual Studio 8\VC\include}).
-
-@item MSVC++ also does not understand the @code{inline} keyword used by
-FFmpeg, so you must add this line before @code{#include}ing libav*:
-@example
-#define inline _inline
-@end example
-
-@item Build your application, everything should work.
-
-@end enumerate
-
-@subsection Using shared libraries
-
-This is how to create DLL and LIB files that are compatible with MSVC++:
-
-@enumerate
-
-@item Add a call to @file{vcvars32.bat} (which sets up the environment
-variables for the Visual C++ tools) as the first line of @file{msys.bat}.
-The standard location for @file{vcvars32.bat} is
-@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat},
-and the standard location for @file{msys.bat} is @file{C:\msys\1.0\msys.bat}.
-If this corresponds to your setup, add the following line as the first line
-of @file{msys.bat}:
+Finally, run:

@example
-call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"
-@end example
+For MSVC:
+./configure --toolchain=msvc

-Alternatively, you may start the @file{Visual Studio 2005 Command Prompt},
-and run @file{c:\msys\1.0\msys.bat} from there.
+For ICL:
+./configure --toolchain=icl

-@item Within the MSYS shell, run @code{lib.exe}. If you get a help message
-from @file{Microsoft (R) Library Manager}, this means your environment
-variables are set up correctly, the @file{Microsoft (R) Library Manager}
-is on the path and will be used by FFmpeg to create
-MSVC++-compatible import libraries.
-
-@item Build FFmpeg with
-
-@example
-./configure --enable-shared
 make
 make install
@end example

-Your install path (@file{/usr/local/} by default) should now have the
-necessary DLL and LIB files under the @file{bin} directory.
+If you wish to compile shared libraries, add @code{--enable-shared} to your
+configure options. Note that due to the way MSVC and ICL handle DLL imports and
+exports, you cannot compile static and shared libraries at the same time, and
+enabling shared libraries will automatically disable the static ones.

+Notes:
+
+@itemize
+
+@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
+You can find out by running @code{which link} to see which @code{link.exe} you
+are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
+in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
+@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
+
+@item If you wish to build with zlib support, you will have to grab a compatible
+zlib binary from somewhere, with an MSVC import lib, or if you wish to link
+statically, you can follow the instructions below to build a compatible
+@code{zlib.lib} with MSVC. Regardless of which method you use, you must still
+follow step 3, or compilation will fail.
+@enumerate
+@item Grab the @uref{http://zlib.net/, zlib sources}.
+@item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since
+this is how FFmpeg is built as well.
+@item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets
+erroneously included when building FFmpeg.
+@item Run @code{nmake -f win32/Makefile.msc}.
+@item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC
+can see.
@end enumerate

-Alternatively, build the libraries with a cross compiler, according to
-the instructions below in @ref{Cross compilation for Windows with Linux}.
+@item FFmpeg has been tested with the following on i686 and x86_64:
+@itemize
+@item Visual Studio 2010 Pro and Express
+@item Visual Studio 2012 Pro and Express
+@item Intel Composer XE 2013
+@end itemize
+Anything else is not officially supported.

-To use those files with MSVC++, do the same as you would do with
-the static libraries, as described above. But in Step 4,
-you should only need to add the directory where the LIB files are installed
-(i.e. @file{c:\msys\usr\local\bin}). This is not a typo, the LIB files are
-installed in the @file{bin} directory. And instead of adding the static
-libraries (@file{libxxx.a} files) you should add the MSVC import libraries
-(@file{avcodec.lib}, @file{avformat.lib}, and
-@file{avutil.lib}). Note that you should not use the GCC import
-libraries (@file{libxxx.dll.a} files), as these will give you undefined
-reference errors. There should be no need for @file{libmingwex.a},
-@file{libgcc.a}, and @file{wsock32.lib}, nor any other external library
-statically linked into the DLLs.
+@end itemize

-FFmpeg headers do not declare global data for Windows DLLs through the usual
-dllexport/dllimport interface. Such data will be exported properly while
-building, but to use them in your MSVC++ code you will have to edit the
-appropriate headers and mark the data as dllimport. For example, in
-libavutil/pixdesc.h you should have:
+@subsection Linking to FFmpeg with Microsoft Visual C++
+
+If you plan to link with MSVC-built static libraries, you will need
+to make sure you have @code{Runtime Library} set to
+@code{Multi-threaded (/MT)} in your project's settings.
+
+You will need to define @code{inline} to something MSVC understands:
@example
-extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
+#define inline __inline
@end example

-Note that using import libraries created by dlltool requires
-the linker optimization option to be set to
-"References: Keep Unreferenced Data (@code{/OPT:NOREF})", otherwise
-the resulting binaries will fail during runtime. This isn't
-required when using import libraries generated by lib.exe.
+Also note, that as stated in @strong{Microsoft Visual C++}, you will need
+an MSVC-compatible @uref{http://code.google.com/p/msinttypes/, inttypes.h}.
+
+If you plan on using import libraries created by dlltool, you must
+set @code{References} to @code{No (/OPT:NOREF)} under the linker optimization
+settings, otherwise the resulting binaries will fail during runtime.
+This is not required when using import libraries generated by @code{lib.exe}.
 This issue is reported upstream at
@url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}.

@@ -272,12 +214,12 @@ To create import libraries that work with the @code{/OPT:REF} option

@enumerate

-@item Open @emph{Visual Studio 2005 Command Prompt}.
+@item Open the @emph{Visual Studio Command Prompt}.

 Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
 which sets up the environment variables for the Visual C++ tools
-(the standard location for this file is
-@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat}).
+(the standard location for this file is something like
+@file{C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat}).

@item Enter the @file{bin} directory where the created LIB and DLL files
 are stored.
@@ -362,4 +304,67 @@ and for a build with shared libraries
 ./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example

+@chapter Plan 9
+
+The native @uref{http://plan9.bell-labs.com/plan9/, Plan 9} compiler
+does not implement all the C99 features needed by FFmpeg so the gcc
+port must be used.  Furthermore, a few items missing from the C
+library and shell environment need to be fixed.
+
+@itemize
+
+@item GNU awk, grep, make, and sed
+
+Working packages of these tools can be found at
+@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}.
+They can be installed with @uref{http://9front.org/, 9front's} @code{pkg}
+utility by setting @code{pkgpath} to
+@code{http://ports2plan9.googlecode.com/files/}.
+
+@item Missing/broken @code{head} and @code{printf} commands
+
+Replacements adequate for building FFmpeg can be found in the
+@code{compat/plan9} directory.  Place these somewhere they will be
+found by the shell.  These are not full implementations of the
+commands and are @emph{not} suitable for general use.
+
+@item Missing C99 @code{stdint.h} and @code{inttypes.h}
+
+Replacement headers are available from
+@url{http://code.google.com/p/plan9front/issues/detail?id=152}.
+
+@item Missing or non-standard library functions
+
+Some functions in the C library are missing or incomplete.  The
+@code{@uref{http://ports2plan9.googlecode.com/files/gcc-apelibs-1207.tbz,
+gcc-apelibs-1207}} package from
+@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}
+includes an updated C library, but installing the full package gives
+unusable executables.  Instead, keep the files from @code{gccbin.tgz}
+under @code{/386/lib/gnu}.  From the @code{libc.a} archive in the
+@code{gcc-apelibs-1207} package, extract the following object files and
+turn them into a library:
+
+@itemize
+@item @code{strerror.o}
+@item @code{strtoll.o}
+@item @code{snprintf.o}
+@item @code{vsnprintf.o}
+@item @code{vfprintf.o}
+@item @code{_IO_getc.o}
+@item @code{_IO_putc.o}
+@end itemize
+
+Use the @code{--extra-libs} option of @code{configure} to inform the
+build system of this library.
+
+@item FPU exceptions enabled by default
+
+Unlike most other systems, Plan 9 enables FPU exceptions by default.
+These must be disabled before calling any FFmpeg functions.  While the
+included tools will do this automatically, other users of the
+libraries must do it themselves.
+
+@end itemize
+
@bye
--- a/doc/print_options.c
+++ b/doc/print_options.c
@@ -39,6 +39,9 @@ static void print_usage(void)

 static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
 {
+    if (!(o->flags & (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM)))
+        return;
+
    printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
    switch (o->type) {
    case AV_OPT_TYPE_BINARY:   printf("hexadecimal string"); break;
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -49,6 +49,16 @@ Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapte
 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
@end example

+@section cache
+
+Caching wrapper for input stream.
+
+Cache the input stream to temporary file. It brings seeking capability to live streams.
+
+@example
+cache:@var{URL}
+@end example
+
@section concat

 Physical concatenation protocol.
@@ -75,6 +85,34 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 Note that you may need to escape the character "|" which is special for
 many shells.

+@section crypto
+
+AES-encrypted stream reading protocol.
+
+The accepted options are:
+@table @option
+@item key
+Set the AES decryption key binary block from given hexadecimal representation.
+
+@item iv
+Set the AES decryption initialization vector binary block from given hexadecimal representation.
+@end table
+
+Accepted URL formats:
+@example
+crypto:@var{URL}
+crypto+@var{URL}
+@end example
+
+@section data
+
+Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
+
+For example, to convert a GIF file given inline with @command{ffmpeg}:
+@example
+ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
+@end example
+
@section file

 File access protocol.
@@ -91,6 +129,40 @@ The ff* tools default to the file protocol, that is a resource
 specified with the name "FILE.mpeg" is interpreted as the URL
 "file:FILE.mpeg".

+@section ftp
+
+FTP (File Transfer Protocol).
+
+Allow to read from or write to remote resources using FTP protocol.
+
+Following syntax is required.
+@example
+ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
+@end example
+
+This protocol accepts the following options.
+
+@table @option
+@item timeout
+Set timeout of socket I/O operations used by the underlying low level
+operation. By default it is set to -1, which means that the timeout is
+not specified.
+
+@item ftp-anonymous-password
+Password used when login as anonymous user. Typically an e-mail address
+should be used.
+
+@item ftp-write-seekable
+Control seekability of connection during encoding. If set to 1 the
+resource is supposed to be seekable, if set to 0 it is assumed not
+to be seekable. Default value is 0.
+@end table
+
+NOTE: Protocol can be used as output, but it is recommended to not do
+it, unless special care is taken (tests, customized server configuration
+etc.). Different FTP servers behave in different way during seek
+operation. ff* tools may produce incomplete content due to server limitations.
+
@section gopher

 Gopher protocol.
@@ -119,6 +191,77 @@ m3u8 files.

 HTTP (Hyper Text Transfer Protocol).

+This protocol accepts the following options.
+
+@table @option
+@item seekable
+Control seekability of connection. If set to 1 the resource is
+supposed to be seekable, if set to 0 it is assumed not to be seekable,
+if set to -1 it will try to autodetect if it is seekable. Default
+value is -1.
+
+@item chunked_post
+If set to 1 use chunked transfer-encoding for posts, default is 1.
+
+@item headers
+Set custom HTTP headers, can override built in default headers. The
+value must be a string encoding the headers.
+
+@item content_type
+Force a content type.
+
+@item user-agent
+Override User-Agent header. If not specified the protocol will use a
+string describing the libavformat build.
+
+@item multiple_requests
+Use persistent connections if set to 1. By default it is 0.
+
+@item post_data
+Set custom HTTP post data.
+
+@item timeout
+Set timeout of socket I/O operations used by the underlying low level
+operation. By default it is set to -1, which means that the timeout is
+not specified.
+
+@item mime_type
+Set MIME type.
+
+@item icy
+If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
+supports this, the metadata has to be retrieved by the application by reading
+the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
+The default is 0.
+
+@item icy_metadata_headers
+If the server supports ICY metadata, this contains the ICY specific HTTP reply
+headers, separated with newline characters.
+
+@item icy_metadata_packet
+If the server supports ICY metadata, and @option{icy} was set to 1, this
+contains the last non-empty metadata packet sent by the server.
+
+@item cookies
+Set the cookies to be sent in future requests. The format of each cookie is the
+same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
+delimited by a newline character.
+@end table
+
+@subsection HTTP Cookies
+
+Some HTTP requests will be denied unless cookie values are passed in with the
+request. The @option{cookies} option allows these cookies to be specified. At
+the very least, each cookie must specify a value along with a path and domain.
+HTTP requests that match both the domain and path will automatically include the
+cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
+by a newline.
+
+The required syntax to play a stream specifying a cookie is:
+@example
+ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
+@end example
+
@section mmst

 MMS (Microsoft Media Server) protocol over TCP.
@@ -472,6 +615,11 @@ To receive a stream in realtime:
 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@end example

+@table @option
+@item stimeout
+Socket IO timeout in micro seconds.
+@end table
+
@section sap

 Session Announcement Protocol (RFC 2974). This is not technically a
@@ -564,6 +712,50 @@ To play back the first stream announced on one the default IPv6 SAP multicast ad
 ffplay sap://[ff0e::2:7ffe]
@end example

+@section sctp
+
+Stream Control Transmission Protocol.
+
+The accepted URL syntax is:
+@example
+sctp://@var{host}:@var{port}[?@var{options}]
+@end example
+
+The protocol accepts the following options:
+@table @option
+@item listen
+If set to any value, listen for an incoming connection. Outgoing connection is done by default.
+
+@item max_streams
+Set the maximum number of streams. By default no limit is set.
+@end table
+
+@section srtp
+
+Secure Real-time Transport Protocol.
+
+The accepted options are:
+@table @option
+@item srtp_in_suite
+@item srtp_out_suite
+Select input and output encoding suites.
+
+Supported values:
+@table @samp
+@item AES_CM_128_HMAC_SHA1_80
+@item SRTP_AES128_CM_HMAC_SHA1_80
+@item AES_CM_128_HMAC_SHA1_32
+@item SRTP_AES128_CM_HMAC_SHA1_32
+@end table
+
+@item srtp_in_params
+@item srtp_out_params
+Set input and output encoding parameters, which are expressed by a
+base64-encoded representation of a binary block. The first 16 bytes of
+this binary block are used as master key, the following 14 bytes are
+used as master salt.
+@end table
+
@section tcp

 Trasmission Control Protocol.
--- a/doc/rate_distortion.txt
+++ b/doc/rate_distortion.txt
@@ -23,7 +23,7 @@ Let's consider the problem of minimizing:

 rate is the filesize
 distortion is the quality
-lambda is a fixed value choosen as a tradeoff between quality and filesize
+lambda is a fixed value chosen as a tradeoff between quality and filesize
 Is this equivalent to finding the best quality for a given max
 filesize? The answer is yes. For each filesize limit there is some lambda
 factor for which minimizing above will get you the best quality (using your
--- a/doc/resampler.texi
+++ b/doc/resampler.texi
@@ -0,0 +1,226 @@
+@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 input 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}.
+This will automatically be chosen when it is not explicitly set.
+
+@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 the center mix level. It is a value expressed in deciBel, and must be
+in the interval [-32,32].
+
+@item slev, surround_mix_level
+Set the surround mix level. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+
+@item lfe_mix_level
+Set LFE mix into non LFE level. It is used when there is a LFE input but no
+LFE output. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+
+@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, this flag forces resampling to be used even when the
+input and output sample rates match.
+@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
+@item lipshitz
+select lipshitz noise shaping dither
+@item shibata
+select shibata noise shaping dither
+@item low_shibata
+select low shibata noise shaping dither
+@item high_shibata
+select high shibata noise shaping dither
+@item f_weighted
+select f-weighted noise shaping dither
+@item modified_e_weighted
+select modified-e-weighted noise shaping dither
+@item improved_e_weighted
+select improved-e-weighted noise shaping dither
+
+@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, and must be in
+the interval [0,30].
+
+@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 first_pts
+For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
+This allows for padding/trimming at the start of stream. By default, no
+assumption is made about the first frame's expected pts, so no padding or
+trimming is done. For example, this could be set to 0 to pad the beginning with
+silence if an audio stream starts after the video stream or to trim any samples
+with a negative pts due to encoder delay.
+
+@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 in the
+interval [2,16], default value is 9.
+
+@item output_sample_bits
+For swr only, set number of used output sample bits for dithering. Must be an integer in the
+interval [0,64], default value is 0, which means it's not used.
+
+@end table
+
+@c man end RESAMPLER OPTIONS
--- a/doc/scaler.texi
+++ b/doc/scaler.texi
@@ -0,0 +1,99 @@
+@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
--- a/doc/swresample.txt
+++ b/doc/swresample.txt
@@ -31,16 +31,16 @@ Special Converter         v
                          v
                        Output

-Planar/Packed convertion is done when needed during sample format convertion
-Every step can be skipped without memcpy when its not needed.
+Planar/Packed conversion is done when needed during sample format conversion.
+Every step can be skipped without memcpy when it is not needed.
 Either Resampling and Rematrixing can be performed first depending on which
-way its faster.
+way it is faster.
 The Buffers are needed for resampling due to resamplng being a process that
 requires future and past data, it thus also introduces inevitably a delay when
 used.
 Internally 32bit float and 16bit int is supported currently, other formats can
-easily be added
+easily be added.
 Externally all sample formats in packed and planar configuration are supported
-Its also trivial to add special converters for common cases
-If only sample format and or packed/planar convertion is needed it
+It's also trivial to add special converters for common cases.
+If only sample format and/or packed/planar conversion is needed, it
 is performed from input to output directly in a single pass with no intermediates.
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .0.git
 .0