avcodec/h264_parser: rewrite the parse_nal_units() loop logic based on h264.c

Fixes Ticket4011 Signed-off-by: Michael Niedermayer <michaelni@gmx.at> (cherry picked from commit 69a9a90d2e) Conflicts: libavcodec/h264_parser.c
avcodec/h264: simplify find_start_code()
2014-11-01 18:27:12 +01:00 · 2014-11-01 18:26:22 +01:00 · 2014-11-01 18:25:47 +01:00 · 2014-11-01 16:12:52 +01:00 · 2014-11-01 16:12:52 +01:00 · 2014-11-01 16:12:52 +01:00
2806 changed files with 253801 additions and 99041 deletions
@@ -6,6 +6,8 @@
 *.dylib
 *.exe
 *.exp
+*.gcda
+*.gcno
 *.h.c
 *.ilk
 *.lib
@@ -24,43 +26,53 @@
 /ffprobe
 /ffserver
 /config.*
-/version.h
+/coverage.info
 /doc/*.1
 /doc/*.3
 /doc/*.html
 /doc/*.pod
+/doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
-/doc/examples/decoding_encoding
-/doc/examples/demuxing
+/doc/doxy/html/
+/doc/examples/avio_reading
+/doc/examples/avcodec
+/doc/examples/demuxing_decoding
+/doc/examples/filter_audio
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
 /doc/examples/pc-uninstalled
+/doc/examples/remuxing
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
+/doc/examples/transcode_aac
 /doc/fate.txt
-/doc/doxy/html/
 /doc/print_options
+/lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
+/libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
 /tests/rotozoom
 /tests/tiny_psnr
+/tests/tiny_ssim
 /tests/videogen
 /tests/vsynth1/
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
+/tools/crypto_bench
 /tools/cws2fws
 /tools/fourcc2pixfmt
 /tools/ffescape
 /tools/ffeval
+/tools/ffhash
 /tools/graph2dot
 /tools/ismindex
 /tools/pktdumper
@@ -68,3 +80,4 @@
 /tools/qt-faststart
 /tools/trasher
 /tools/seek_print
+/tools/zmqsend
@@ -1,7 +1,172 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

-version <next>:
+version 2.2.7
+- snow: fix null pointer dereference
+- iff: fix out of array access
+- svq1dec: fix input data corruption
+- proresenc_ks: check buffer size
+
+
+version 2.2.6
+- fix infinite loop in dvbsub parser
+- fix some interlaced MPEG-2 videos
+- fix decoding issues in dv (Ticket2340, 2341)
+- fix v4l2 and v4l2enc crashes
+- fix theoretical librtmp crash
+- fix theoretical eamad crash
+- support dimension change in g2meet
+
+
+version 2.2:
+- HNM version 4 demuxer and video decoder
+- Live HDS muxer
+- setsar/setdar filters now support variables in ratio expressions
+- elbg filter
+- string validation in ffprobe
+- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
+- complete Voxware MetaSound decoder
+- remove mp3_header_compress bitstream filter
+- Windows resource files for shared libraries
+- aeval filter
+- stereoscopic 3d metadata handling
+- WebP encoding via libwebp
+- ATRAC3+ decoder
+- VP8 in Ogg demuxing
+- side & metadata support in NUT
+- framepack filter
+- XYZ12 rawvideo support in NUT
+- Exif metadata support in WebP decoder
+- OpenGL device
+- Use metadata_header_padding to control padding in ID3 tags (currently used in
+  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
+- Mirillis FIC video decoder
+- Support DNx444
+- libx265 encoder
+- dejudder filter
+- Autodetect VDA like all other hardware accelerations
+
+
+version 2.1:
+
+- aecho filter
+- perspective filter ported from libmpcodecs
+- ffprobe -show_programs option
+- compand filter
+- RTMP seek support
+- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
+  even when used as an input option. Previous behavior can be restored with
+  the -noaccurate_seek option.
+- ffmpeg -t option can now be used for inputs, to limit the duration of
+  data read from an input file
+- incomplete Voxware MetaSound decoder
+- read EXIF metadata from JPEG
+- DVB teletext decoder
+- phase filter ported from libmpcodecs
+- w3fdif filter
+- Opus support in Matroska
+- FFV1 version 1.3 is stable and no longer experimental
+- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
+- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
+  more consistent with other muxers.
+- adelay filter
+- pullup filter ported from libmpcodecs
+- ffprobe -read_intervals option
+- Lossless and alpha support for WebP decoder
+- Error Resilient AAC syntax (ER AAC LC) decoding
+- Low Delay AAC (ER AAC LD) decoding
+- mux chapters in ASF files
+- SFTP protocol (via libssh)
+- libx264: add ability to encode in YUVJ422P and YUVJ444P
+- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
+- make decoding alpha optional for prores, ffv1 and vp6 by setting
+  the skip_alpha flag.
+- ladspa wrapper filter
+- native VP9 decoder
+- dpx parser
+- max_error_rate parameter in ffmpeg
+- PulseAudio output device
+- ReplayGain scanner
+- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
+- Linux framebuffer output device
+- HEVC decoder
+- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
+- mergeplanes filter
+
+
+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
+- 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
+- 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
@@ -535,7 +700,7 @@ version 0.6:
 - LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
 - WMA Pro decoder
 - Core Audio Format demuxer
- Atrac1 decoder
+- ATRAC1 decoder
 - MD STUDIO audio demuxer
 - RF64 support in WAV demuxer
 - MPEG-4 Audio Lossless Coding (ALS) decoder
@@ -635,7 +800,7 @@ version 0.5:
 - MXF demuxer
 - VC-1/WMV3/WMV9 video decoder
 - MacIntel support
- AVISynth support
+- AviSynth support
 - VMware video decoder
 - VP5 video decoder
 - VP6 video decoder
@@ -663,7 +828,7 @@ version 0.5:
 - Interplay C93 demuxer and video decoder
 - Bethsoft VID demuxer and video decoder
 - CRYO APC demuxer
- Atrac3 decoder
+- ATRAC3 decoder
 - V.Flash PTX decoder
 - RoQ muxer, RoQ audio encoder
 - Renderware TXD demuxer and decoder
@@ -33,16 +33,21 @@ Specifically, the GPL parts of FFmpeg are
    - vf_geq.c
    - vf_histeq.c
    - vf_hqdn3d.c
-    - vf_hue.c
+    - vf_interlace.c
    - vf_kerndeint.c
+    - vf_mcdeint.c
    - vf_mp.c
-    - vf_noise.c
+    - vf_owdenoise.c
+    - vf_perspective.c
+    - vf_phase.c
    - vf_pp.c
+    - vf_pullup.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
@@ -69,8 +74,16 @@ 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
+    - libx265
+    - 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
@@ -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
@@ -31,7 +31,7 @@ ffprobe:
  ffprobe.c                             Stefano Sabatini

 ffserver:
-  ffserver.c, ffserver.h                Baptiste Coudurier
+  ffserver.c                            Reynaldo H. Verdejo Pinochet

 Commandline utility code:
  cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -43,16 +43,24 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================

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


+Communication
+=============
+
+website                                 Robert Swain, Lou Logan
+mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
+Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
+Twitter                                 Lou Logan
+Launchpad                               Timothy Gu
+
+
 libavutil
 =========

@@ -62,11 +70,23 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer

 Other:
-  intfloat*                             Michael Niedermayer
-  rational.c, rational.h                Michael Niedermayer
-  mathematics.c, mathematics.h          Michael Niedermayer
-  integer.c, integer.h                  Michael Niedermayer
+  bprint                                Nicolas George
  bswap.h
+  des                                   Reimar Doeffinger
+  eval.c, eval.h                        Michael Niedermayer
+  float_dsp                             Loren Merritt
+  hash                                  Reimar Doeffinger
+  intfloat*                             Michael Niedermayer
+  integer.c, integer.h                  Michael Niedermayer
+  lzo                                   Reimar Doeffinger
+  mathematics.c, mathematics.h          Michael Niedermayer
+  mem.c, mem.h                          Michael Niedermayer
+  opencl.c, opencl.h                    Wei Gao
+  opt.c, opt.h                          Michael Niedermayer
+  rational.c, rational.h                Michael Niedermayer
+  rc4                                   Reimar Doeffinger
+  ripemd.c, ripemd.h                    James Almer
+  timecode                              Clément Bœsch


 libavcodec
@@ -77,10 +97,6 @@ Generic Parts:
    avcodec.h                           Michael Niedermayer
  utility code:
    utils.c                             Michael Niedermayer
-    mem.c                               Michael Niedermayer
-    opt.c, opt.h                        Michael Niedermayer
-  arithmetic expression evaluator:
-    eval.c                              Michael Niedermayer
  audio and video frame extraction:
    parser.c                            Michael Niedermayer
  bitstream reading:
@@ -111,6 +127,8 @@ Generic Parts:
    libpostproc/*                       Michael Niedermayer
  table generation:
    tableprint.c, tableprint.h          Reimar Doeffinger
+  fixed point FFT:
+    fft*                                Zeljko Lukac

 Codecs:
  4xm.c                                 Michael Niedermayer
@@ -124,14 +142,16 @@ Codecs:
  ass*                                  Aurelien Jacobs
  asv*                                  Michael Niedermayer
  atrac3*                               Benjamin Larsson
+  atrac3plus*                           Maxim Poliakovski
  bgmc.c, bgmc.h                        Thilo Borgmann
  bink.c                                Kostya Shishkov
  binkaudio.c                           Peter Ross
  bmp.c                                 Mans Rullgard, Kostya Shishkov
  cavs*                                 Stefan Gehrer
-  celp_filters.*                        Vitor Sessak
  cdxl.c                                Paul B Mahol
+  celp_filters.*                        Vitor Sessak
  cinepak.c                             Roberto Togni
+  cinepakenc.c                          Rl / Aetey G.T. AB
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
  cook.c, cookdata.h                    Benjamin Larsson
@@ -141,9 +161,10 @@ Codecs:
  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
-  dxa.c                                 Kostya Shishkov
  dv.c                                  Roman Shaposhnik
+  dxa.c                                 Kostya Shishkov
  eacmv*, eaidct*, eat*                 Peter Ross
+  exif.c, exif.h                        Thilo Borgmann
  ffv1.c                                Michael Niedermayer
  ffwavesynth.c                         Nicolas George
  flac*                                 Justin Ruggles
@@ -152,9 +173,9 @@ Codecs:
  g722.c                                Martin Storsjo
  g726.c                                Roman Shaposhnik
  gifdec.c                              Baptiste Coudurier
-  h264*                                 Loren Merritt, Michael Niedermayer
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
+  h264*                                 Loren Merritt, Michael Niedermayer
  huffyuv.c                             Michael Niedermayer
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
@@ -163,13 +184,14 @@ 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
  lcl*.c                                Roberto Togni, Reimar Doeffinger
  libcelt_dec.c                         Nicolas George
-  libgsm.c                              Michel Bardiaux
  libdirac*                             David Conrad
+  libgsm.c                              Michel Bardiaux
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
@@ -177,8 +199,11 @@ Codecs:
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
-  libxavs.c                             Stefan Gehrer
+  libvpx*                               James Zern
  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
+  libx265.c                             Derek Buitenhuis
+  libxavs.c                             Stefan Gehrer
+  libzvbi-teletextdec.c                 Marton Balint
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
@@ -189,6 +214,7 @@ Codecs:
  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
@@ -214,6 +240,7 @@ 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
@@ -227,6 +254,7 @@ 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
@@ -237,12 +265,13 @@ Codecs:
  vda_h264_dec.c                        Xidorn Quan
  vima.c                                Paul B Mahol
  vmnc.c                                Kostya Shishkov
-  vorbis_enc.c                          Oded Shimon
  vorbis_dec.c                          Denes Balatoni, David Conrad
+  vorbis_enc.c                          Oded Shimon
  vp3*                                  Mike Melanson
  vp5                                   Aurelien Jacobs
  vp6                                   Aurelien Jacobs
  vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
+  vp9                                   Ronald Bultje, Clément Bœsch
  vqavideo.c                            Mike Melanson
  wavpack.c                             Kostya Shishkov
  wmaprodec.c                           Sascha Sommer
@@ -251,6 +280,7 @@ Codecs:
  wnv1.c                                Kostya Shishkov
  xan.c                                 Mike Melanson
  xbm*                                  Paul B Mahol
+  xface                                 Stefano Sabatini
  xl.c                                  Kostya Shishkov
  xvmc.c                                Ivan Kalvachev
  xwd*                                  Paul B Mahol
@@ -272,11 +302,16 @@ libavdevice
    libavdevice/avdevice.h


+  dshow.c                               Roger Pack
+  fbdev_enc.c                           Lukasz Marek
  iec61883.c                            Georg Lippitsch
+  lavfi                                 Stefano Sabatini
  libdc1394.c                           Roman Shaposhnik
+  opengl_enc.c                          Lukasz Marek
+  pulse_audio_enc.c                     Lukasz Marek
+  sdl                                   Stefano Sabatini
  v4l2.c                                Luca Abeni
  vfwcap.c                              Ramiro Polla
-  dshow.c                               Roger Pack

 libavfilter
 ===========
@@ -285,10 +320,34 @@ Generic parts:
  graphdump.c                           Nicolas George

 Filters:
+  af_adelay.c                           Paul B Mahol
+  af_aecho.c                            Paul B Mahol
+  af_afade.c                            Paul B Mahol
  af_amerge.c                           Nicolas George
+  af_aphaser.c                          Paul B Mahol
+  af_aresample.c                        Michael Niedermayer
+  af_astats.c                           Paul B Mahol
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
+  af_biquads.c                          Paul B Mahol
+  af_compand.c                          Paul B Mahol
+  af_ladspa.c                           Paul B Mahol
  af_pan.c                              Nicolas George
+  avf_avectorscope.c                    Paul B Mahol
+  vf_blend.c                            Paul B Mahol
+  vf_colorbalance.c                     Paul B Mahol
+  vf_dejudder.c                         Nicholas Robbins
+  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
+  vf_drawbox.c/drawgrid                 Andrey Utkin
+  vf_extractplanes.c                    Paul B Mahol
+  vf_histogram.c                        Paul B Mahol
+  vf_il.c                               Paul B Mahol
+  vf_mergeplanes.c                      Paul B Mahol
+  vf_psnr.c                             Paul B Mahol
+  vf_scale.c                            Michael Niedermayer
+  vf_separatefields.c                   Paul B Mahol
+  vf_stereo3d.c                         Paul B Mahol
+  vf_telecine.c                         Paul B Mahol
  vf_yadif.c                            Michael Niedermayer

 Sources:
@@ -308,12 +367,14 @@ Muxers/Demuxers:
  4xm.c                                 Mike Melanson
  adtsenc.c                             Robert Swain
  afc.c                                 Paul B Mahol
-  aiff.c                                Baptiste Coudurier
+  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
+  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
  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
@@ -332,11 +393,12 @@ Muxers/Demuxers:
  flvdec.c, flvenc.c                    Michael Niedermayer
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
+  hls.c                                 Anssi Hannula
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
-  ipmovie.c                             Mike Melanson
  img2*.c                               Michael Niedermayer
+  ipmovie.c                             Mike Melanson
  ircam*                                Paul B Mahol
  iss.c                                 Stefan Gehrer
  jacosub*                              Clément Bœsch
@@ -350,11 +412,11 @@ Muxers/Demuxers:
  matroskadec.c                         Aurelien Jacobs
  matroskaenc.c                         David Conrad
  metadata*                             Aurelien Jacobs
-  microdvd*                             Aurelien Jacobs
  mgsts.c                               Paul B Mahol
+  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
  mov.c                                 Michael Niedermayer, Baptiste Coudurier
-  movenc.c                              Michael Niedermayer, Baptiste Coudurier
+  movenc.c                              Baptiste Coudurier, Matthieu Bouron
  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
  mpegenc.c                             Michael Niedermayer
@@ -391,6 +453,7 @@ Muxers/Demuxers:
  siff.c                                Kostya Shishkov
  smacker.c                             Kostya Shishkov
  smjpeg*                               Paul B Mahol
+  spdif*                                Anssi Hannula
  srtdec.c                              Aurelien Jacobs
  swf.c                                 Baptiste Coudurier
  takdec.c                              Paul B Mahol
@@ -399,6 +462,7 @@ Muxers/Demuxers:
  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
@@ -406,7 +470,9 @@ Muxers/Demuxers:

 Protocols:
  bluray.c                              Petri Hintukainen
+  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
+  libssh.c                              Lukasz Marek
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni

@@ -431,7 +497,7 @@ Operating systems / CPU architectures
 Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
-MIPS                                    Mans Rullgard
+MIPS                                    Mans Rullgard, Nedeljko Babic
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
@@ -445,17 +511,20 @@ x86                                     Michael Niedermayer
 Releases
 ========

+2.2                                     Michael Niedermayer
+2.1                                     Michael Niedermayer
 1.2                                     Michael Niedermayer
-1.1                                     Michael Niedermayer
-1.0                                     Michael Niedermayer

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


 GnuPG Fingerprints of maintainers and contributors
 ==================================================

+Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 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
@@ -463,8 +532,10 @@ 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
@@ -481,3 +552,4 @@ Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
@@ -6,30 +6,34 @@ vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
+vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)

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

-PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS      += $(AVPROGS)

-OBJS        = cmdutils.o $(EXEOBJS)
-OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
+AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
+ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
+
+OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
+OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher
+TOOLS       = qt-faststart trasher uncoded_frame
 TOOLS-$(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-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
@@ -44,23 +48,21 @@ FFLIBS := avutil
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
 EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README

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

 include $(SRC_PATH)/common.mak

 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)

-all: $(PROGS)
-
-$(PROGS): %$(EXESUF): %_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+all: $(AVPROGS)

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

 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
+tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
+tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)

 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -74,7 +76,7 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
               ALTIVEC-OBJS VIS-OBJS                                     \
               MMX-OBJS YASM-OBJS                                        \
               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               OBJS HOSTOBJS TESTOBJS
+               OBJS SLIBOBJS HOSTOBJS TESTOBJS

 define RESET
 $(1) :=
@@ -91,8 +93,10 @@ endef

 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))

+include $(SRC_PATH)/doc/Makefile
+
 define DOPROG
-OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
+OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
@@ -100,7 +104,13 @@ $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef

-$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
+$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
+
+ffprobe.o cmdutils.o : libavutil/ffversion.h
+
+$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@

 %$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
@@ -115,14 +125,14 @@ GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@

-version.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
+libavutil/ffversion.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
 	$(Q)touch .version

 # force version.sh to run whenever version might have changed
 -include .version

-ifdef PROGS
+ifdef AVPROGS
 install: install-progs install-data
 endif

@@ -133,7 +143,7 @@ install-libs: install-libs-yes
 install-progs-yes:
 install-progs-$(CONFIG_SHARED): install-libs

-install-progs: install-progs-yes $(PROGS)
+install-progs: install-progs-yes $(AVPROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"

@@ -145,37 +155,27 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data

 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))

 uninstall-data:
 	$(RM) -r "$(DATADIR)"

 clean::
-	$(RM) $(ALLPROGS) $(ALLPROGS_G)
+	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(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 libavutil/ffversion.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
 include $(SRC_PATH)/tests/Makefile

 $(sort $(OBJDIRS)):
@@ -1 +1 @@
-1.1.git
+2.2.10
@@ -20,6 +20,7 @@
 */

 #include <string.h>
+#include <stdint.h>
 #include <stdlib.h>
 #include <errno.h>
 #include <math.h>
@@ -36,9 +37,7 @@
 #include "libavresample/avresample.h"
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
-#if CONFIG_POSTPROC
 #include "libpostproc/postprocess.h"
-#endif
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
@@ -49,8 +48,9 @@
 #include "libavutil/eval.h"
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
+#include "libavutil/cpu.h"
+#include "libavutil/ffversion.h"
 #include "cmdutils.h"
-#include "version.h"
 #if CONFIG_NETWORK
 #include "libavformat/network.h"
 #endif
@@ -65,9 +65,8 @@ struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;

-const int this_year = 2013;
-
 static FILE *report_file;
+int hide_banner = 0;

 void init_opts(void)
 {
@@ -109,6 +108,21 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
    fflush(report_file);
 }

+static void (*program_exit)(int ret);
+
+void register_exit(void (*cb)(int ret))
+{
+    program_exit = cb;
+}
+
+void exit_program(int ret)
+{
+    if (program_exit)
+        program_exit(ret);
+
+    exit(ret);
+}
+
 double parse_number_or_die(const char *context, const char *numstr, int type,
                           double min, double max)
 {
@@ -126,7 +140,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
    else
        return d;
    av_log(NULL, AV_LOG_FATAL, error, context, numstr, min, max);
-    exit(1);
+    exit_program(1);
    return 0;
 }

@@ -137,7 +151,7 @@ int64_t parse_time_or_die(const char *context, const char *timestr,
    if (av_parse_time(&us, timestr, is_duration) < 0) {
        av_log(NULL, AV_LOG_FATAL, "Invalid %s specification for %s: %s\n",
               is_duration ? "duration" : "date", context, timestr);
-        exit(1);
+        exit_program(1);
    }
    return us;
 }
@@ -196,7 +210,10 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
    return po;
 }

-#if HAVE_COMMANDLINETOARGVW
+/* _WIN32 means using the windows libc - cygwin doesn't define that
+ * by default. HAVE_COMMANDLINETOARGVW is true on cygwin, while
+ * it doesn't provide the actual command line via GetCommandLineW(). */
+#if HAVE_COMMANDLINETOARGVW && defined(_WIN32)
 #include <windows.h>
 #include <shellapi.h>
 /* Will be leaked on exit */
@@ -280,7 +297,7 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
    if (po->flags & OPT_STRING) {
        char *str;
        str = av_strdup(arg);
-//         av_freep(dst);
+        av_freep(dst);
        *(char **)dst = str;
    } else if (po->flags & OPT_BOOL || po->flags & OPT_INT) {
        *(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
@@ -296,12 +313,13 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
        int ret = po->u.func_arg(optctx, opt, arg);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR,
-                   "Failed to set value '%s' for option '%s'\n", arg, opt);
+                   "Failed to set value '%s' for option '%s': %s\n",
+                   arg, opt, av_err2str(ret));
            return ret;
        }
    }
    if (po->flags & OPT_EXIT)
-        exit(0);
+        exit_program(0);

    return 0;
 }
@@ -361,7 +379,7 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
            opt++;

            if ((ret = parse_option(optctx, opt, argv[optindex], options)) < 0)
-                exit(1);
+                exit_program(1);
            optindex += ret;
        } else {
            if (parse_arg_function)
@@ -380,6 +398,16 @@ int parse_optgroup(void *optctx, OptionGroup *g)
    for (i = 0; i < g->nb_opts; i++) {
        Option *o = &g->opts[i];

+        if (g->group_def->flags &&
+            !(g->group_def->flags & o->opt->flags)) {
+            av_log(NULL, AV_LOG_ERROR, "Option %s (%s) cannot be applied to "
+                   "%s %s -- you are trying to apply an input option to an "
+                   "output file or vice versa. Move this option before the "
+                   "file it belongs to.\n", o->key, o->opt->help,
+                   g->group_def->name, g->arg);
+            return AVERROR(EINVAL);
+        }
+
        av_log(NULL, AV_LOG_DEBUG, "Applying option %s (%s) with argument %s.\n",
               o->key, o->opt->help, o->val);

@@ -464,6 +492,18 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
            fflush(report_file);
        }
    }
+    idx = locate_option(argc, argv, options, "hide_banner");
+    if (idx)
+        hide_banner = 1;
+}
+
+static const AVOption *opt_find(void *obj, const char *name, const char *unit,
+                            int opt_flags, int search_flags)
+{
+    const AVOption *o = av_opt_find(obj, name, unit, opt_flags, search_flags);
+    if(o && !o->flags)
+        return NULL;
+    return o;
 }

 #define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
@@ -486,23 +526,23 @@ int opt_default(void *optctx, const char *opt, const char *arg)
        p = opt + strlen(opt);
    av_strlcpy(opt_stripped, opt, FFMIN(sizeof(opt_stripped), p - opt + 1));

-    if ((o = av_opt_find(&cc, opt_stripped, NULL, 0,
+    if ((o = opt_find(&cc, opt_stripped, NULL, 0,
                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) ||
        ((opt[0] == 'v' || opt[0] == 'a' || opt[0] == 's') &&
-         (o = av_opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
+         (o = opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
        av_dict_set(&codec_opts, opt, arg, FLAGS);
        consumed = 1;
    }
-    if ((o = av_opt_find(&fc, opt, NULL, 0,
-                              AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+    if ((o = opt_find(&fc, opt, NULL, 0,
+                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        av_dict_set(&format_opts, opt, arg, FLAGS);
-        if(consumed)
-            av_log(NULL, AV_LOG_VERBOSE, "Routing %s to codec and muxer layer\n", opt);
+        if (consumed)
+            av_log(NULL, AV_LOG_VERBOSE, "Routing option %s to both codec and muxer layer\n", opt);
        consumed = 1;
    }
 #if CONFIG_SWSCALE
    sc = sws_get_class();
-    if (!consumed && av_opt_find(&sc, opt, NULL, 0,
+    if (!consumed && opt_find(&sc, opt, NULL, 0,
                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
        // XXX we only support sws_flags, not arbitrary sws options
        int ret = av_opt_set(sws_opts, opt, arg, 0);
@@ -515,7 +555,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
 #endif
 #if CONFIG_SWRESAMPLE
    swr_class = swr_get_class();
-    if (!consumed && (o=av_opt_find(&swr_class, opt, NULL, 0,
+    if (!consumed && (o=opt_find(&swr_class, opt, NULL, 0,
                                    AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        struct SwrContext *swr = swr_alloc();
        int ret = av_opt_set(swr, opt, arg, 0);
@@ -529,7 +569,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
    }
 #endif
 #if CONFIG_AVRESAMPLE
-    if ((o=av_opt_find(&rc, opt, NULL, 0,
+    if ((o=opt_find(&rc, opt, NULL, 0,
                       AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        av_dict_set(&resample_opts, opt, arg, FLAGS);
        consumed = 1;
@@ -624,7 +664,7 @@ static void init_parse_context(OptionParseContext *octx,
    octx->nb_groups = nb_groups;
    octx->groups    = av_mallocz(sizeof(*octx->groups) * octx->nb_groups);
    if (!octx->groups)
-        exit(1);
+        exit_program(1);

    for (i = 0; i < octx->nb_groups; i++)
        octx->groups[i].group_def = &groups[i];
@@ -768,6 +808,18 @@ do {                                                                           \
    return 0;
 }

+int opt_cpuflags(void *optctx, const char *opt, const char *arg)
+{
+    int ret;
+    unsigned flags = av_get_cpu_flags();
+
+    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
+        return ret;
+
+    av_force_cpu_flags(flags);
+    return 0;
+}
+
 int opt_loglevel(void *optctx, const char *opt, const char *arg)
 {
    const struct { const char *name; int level; } log_levels[] = {
@@ -784,6 +836,13 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
    int level;
    int i;

+    tail = strstr(arg, "repeat");
+    av_log_set_flags(tail ? 0 : AV_LOG_SKIP_REPEATED);
+    if (tail == arg)
+        arg += 6 + (arg[6]=='+');
+    if(tail && !*arg)
+        return 0;
+
    for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) {
        if (!strcmp(log_levels[i].name, arg)) {
            av_log_set_level(log_levels[i].level);
@@ -797,7 +856,7 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
               "Possible levels are numbers or:\n", arg);
        for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++)
            av_log(NULL, AV_LOG_FATAL, "\"%s\"\n", log_levels[i].name);
-        exit(1);
+        exit_program(1);
    }
    av_log_set_level(level);
    return 0;
@@ -890,7 +949,6 @@ static int init_report(const char *env)
           tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday,
           tm->tm_hour, tm->tm_min, tm->tm_sec,
           filename.str);
-    av_log_set_level(FFMAX(av_log_get_level(), AV_LOG_VERBOSE));
    av_bprint_finalize(&filename, NULL);
    return 0;
 }
@@ -908,24 +966,12 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg)
    max = strtol(arg, &tail, 10);
    if (*tail) {
        av_log(NULL, AV_LOG_FATAL, "Invalid max_alloc \"%s\".\n", arg);
-        exit(1);
+        exit_program(1);
    }
    av_max_alloc(max);
    return 0;
 }

-int opt_cpuflags(void *optctx, const char *opt, const char *arg)
-{
-    int ret;
-    unsigned flags = av_get_cpu_flags();
-
-    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
-        return ret;
-
-    av_force_cpu_flags(flags);
-    return 0;
-}
-
 int opt_timelimit(void *optctx, const char *opt, const char *arg)
 {
 #if HAVE_SETRLIMIT
@@ -991,12 +1037,10 @@ static void print_all_libs_info(int flags, int level)
    PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
    PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
    PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
-//    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
+    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
    PRINT_LIB_INFO(swscale,  SWSCALE,  flags, level);
    PRINT_LIB_INFO(swresample,SWRESAMPLE,  flags, level);
-#if CONFIG_POSTPROC
    PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
-#endif
 }

 static void print_program_info(int flags, int level)
@@ -1006,7 +1050,7 @@ static void print_program_info(int flags, int level)
    av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
    if (flags & SHOW_COPYRIGHT)
        av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
-               program_birth_year, this_year);
+               program_birth_year, CONFIG_THIS_YEAR);
    av_log(NULL, level, "\n");
    av_log(NULL, level, "%sbuilt on %s %s with %s\n",
           indent, __DATE__, __TIME__, CC_IDENT);
@@ -1014,10 +1058,36 @@ static void print_program_info(int flags, int level)
    av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }

+static void print_buildconf(int flags, int level)
+{
+    const char *indent = flags & INDENT ? "  " : "";
+    char str[] = { FFMPEG_CONFIGURATION };
+    char *conflist, *remove_tilde, *splitconf;
+
+    // Change all the ' --' strings to '~--' so that
+    // they can be identified as tokens.
+    while ((conflist = strstr(str, " --")) != NULL) {
+        strncpy(conflist, "~--", 3);
+    }
+
+    // Compensate for the weirdness this would cause
+    // when passing 'pkg-config --static'.
+    while ((remove_tilde = strstr(str, "pkg-config~")) != NULL) {
+        strncpy(remove_tilde, "pkg-config ", 11);
+    }
+
+    splitconf = strtok(str, "~");
+    av_log(NULL, level, "\n%sconfiguration:\n", indent);
+    while (splitconf != NULL) {
+        av_log(NULL, level, "%s%s%s\n", indent, indent, splitconf);
+        splitconf = strtok(NULL, "~");
+    }
+}
+
 void show_banner(int argc, char **argv, const OptionDef *options)
 {
    int idx = locate_option(argc, argv, options, "version");
-    if (idx)
+    if (hide_banner || idx)
        return;

    print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
@@ -1034,6 +1104,14 @@ int show_version(void *optctx, const char *opt, const char *arg)
    return 0;
 }

+int show_buildconf(void *optctx, const char *opt, const char *arg)
+{
+    av_log_set_callback(log_callback_help);
+    print_buildconf      (INDENT|0, AV_LOG_INFO);
+
+    return 0;
+}
+
 int show_license(void *optctx, const char *opt, const char *arg)
 {
 #if CONFIG_NONFREE
@@ -1176,7 +1254,8 @@ static void print_codec(const AVCodec *c)
    printf("%s %s [%s]:\n", encoder ? "Encoder" : "Decoder", c->name,
           c->long_name ? c->long_name : "");

-    if (c->type == AVMEDIA_TYPE_VIDEO) {
+    if (c->type == AVMEDIA_TYPE_VIDEO ||
+        c->type == AVMEDIA_TYPE_AUDIO) {
        printf("    Threading capabilities: ");
        switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
                                   CODEC_CAP_SLICE_THREADS)) {
@@ -1257,7 +1336,7 @@ static unsigned get_codecs_sorted(const AVCodecDescriptor ***rcodecs)
        nb_codecs++;
    if (!(codecs = av_calloc(nb_codecs, sizeof(*codecs)))) {
        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
-        exit(1);
+        exit_program(1);
    }
    desc = NULL;
    while ((desc = avcodec_descriptor_next(desc)))
@@ -1410,36 +1489,62 @@ int show_protocols(void *optctx, const char *opt, const char *arg)

 int show_filters(void *optctx, const char *opt, const char *arg)
 {
-    AVFilter av_unused(**filter) = NULL;
+    const AVFilter av_unused(*filter) = NULL;
    char descr[64], *descr_cur;
    int i, j;
    const AVFilterPad *pad;

-    printf("Filters:\n");
+    printf("Filters:\n"
+           "  T.. = Timeline support\n"
+           "  .S. = Slice threading\n"
+           "  ..C = Commmand support\n"
+           "  A = Audio input/output\n"
+           "  V = Video input/output\n"
+           "  N = Dynamic number and/or type of input/output\n"
+           "  | = Source or sink filter\n");
 #if CONFIG_AVFILTER
-    while ((filter = av_filter_next(filter)) && *filter) {
+    while ((filter = avfilter_next(filter))) {
        descr_cur = descr;
        for (i = 0; i < 2; i++) {
            if (i) {
                *(descr_cur++) = '-';
                *(descr_cur++) = '>';
            }
-            pad = i ? (*filter)->outputs : (*filter)->inputs;
+            pad = i ? filter->outputs : filter->inputs;
            for (j = 0; pad && pad[j].name; j++) {
                if (descr_cur >= descr + sizeof(descr) - 4)
                    break;
                *(descr_cur++) = get_media_type_char(pad[j].type);
            }
            if (!j)
-                *(descr_cur++) = '|';
+                *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
+                                  ( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|';
        }
        *descr_cur = 0;
-        printf("%-16s %-10s %s\n", (*filter)->name, descr, (*filter)->description);
+        printf(" %c%c%c %-16s %-10s %s\n",
+               filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.',
+               filter->flags & AVFILTER_FLAG_SLICE_THREADS    ? 'S' : '.',
+               filter->process_command                        ? 'C' : '.',
+               filter->name, descr, filter->description);
    }
 #endif
    return 0;
 }

+int show_colors(void *optctx, const char *opt, const char *arg)
+{
+    const char *name;
+    const uint8_t *rgb;
+    int i;
+
+    printf("%-32s #RRGGBB\n", "name");
+
+    for (i = 0; name = av_get_known_color_name(i, &rgb); i++)
+        printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]);
+
+    return 0;
+}
+
 int show_pix_fmts(void *optctx, const char *opt, const char *arg)
 {
    const AVPixFmtDescriptor *pix_desc = NULL;
@@ -1461,11 +1566,11 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg)
    while ((pix_desc = av_pix_fmt_desc_next(pix_desc))) {
        enum AVPixelFormat pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
        printf("%c%c%c%c%c %-16s       %d            %2d\n",
-               sws_isSupportedInput (pix_fmt)      ? 'I' : '.',
-               sws_isSupportedOutput(pix_fmt)      ? 'O' : '.',
-               pix_desc->flags & PIX_FMT_HWACCEL   ? 'H' : '.',
-               pix_desc->flags & PIX_FMT_PAL       ? 'P' : '.',
-               pix_desc->flags & PIX_FMT_BITSTREAM ? 'B' : '.',
+               sws_isSupportedInput (pix_fmt)              ? 'I' : '.',
+               sws_isSupportedOutput(pix_fmt)              ? 'O' : '.',
+               pix_desc->flags & AV_PIX_FMT_FLAG_HWACCEL   ? 'H' : '.',
+               pix_desc->flags & AV_PIX_FMT_FLAG_PAL       ? 'P' : '.',
+               pix_desc->flags & AV_PIX_FMT_FLAG_BITSTREAM ? 'B' : '.',
               pix_desc->name,
               pix_desc->nb_components,
               av_get_bits_per_pixel(pix_desc));
@@ -1597,6 +1702,62 @@ static void show_help_muxer(const char *name)
        show_help_children(fmt->priv_class, AV_OPT_FLAG_ENCODING_PARAM);
 }

+#if CONFIG_AVFILTER
+static void show_help_filter(const char *name)
+{
+#if CONFIG_AVFILTER
+    const AVFilter *f = avfilter_get_by_name(name);
+    int i, count;
+
+    if (!name) {
+        av_log(NULL, AV_LOG_ERROR, "No filter name specified.\n");
+        return;
+    } else if (!f) {
+        av_log(NULL, AV_LOG_ERROR, "Unknown filter '%s'.\n", name);
+        return;
+    }
+
+    printf("Filter %s\n", f->name);
+    if (f->description)
+        printf("  %s\n", f->description);
+
+    if (f->flags & AVFILTER_FLAG_SLICE_THREADS)
+        printf("    slice threading supported\n");
+
+    printf("    Inputs:\n");
+    count = avfilter_pad_count(f->inputs);
+    for (i = 0; i < count; i++) {
+        printf("       #%d: %s (%s)\n", i, avfilter_pad_get_name(f->inputs, i),
+               media_type_string(avfilter_pad_get_type(f->inputs, i)));
+    }
+    if (f->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)
+        printf("        dynamic (depending on the options)\n");
+    else if (!count)
+        printf("        none (source filter)\n");
+
+    printf("    Outputs:\n");
+    count = avfilter_pad_count(f->outputs);
+    for (i = 0; i < count; i++) {
+        printf("       #%d: %s (%s)\n", i, avfilter_pad_get_name(f->outputs, i),
+               media_type_string(avfilter_pad_get_type(f->outputs, i)));
+    }
+    if (f->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS)
+        printf("        dynamic (depending on the options)\n");
+    else if (!count)
+        printf("        none (sink filter)\n");
+
+    if (f->priv_class)
+        show_help_children(f->priv_class, AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_FILTERING_PARAM |
+                                          AV_OPT_FLAG_AUDIO_PARAM);
+    if (f->flags & AVFILTER_FLAG_SUPPORT_TIMELINE)
+        printf("This filter has support for timeline through the 'enable' option.\n");
+#else
+    av_log(NULL, AV_LOG_ERROR, "Build without libavfilter; "
+           "can not to satisfy request\n");
+#endif
+}
+#endif
+
 int show_help(void *optctx, const char *opt, const char *arg)
 {
    char *topic, *par;
@@ -1617,6 +1778,10 @@ int show_help(void *optctx, const char *opt, const char *arg)
        show_help_demuxer(par);
    } else if (!strcmp(topic, "muxer")) {
        show_help_muxer(par);
+#if CONFIG_AVFILTER
+    } else if (!strcmp(topic, "filter")) {
+        show_help_filter(par);
+#endif
    } else {
        show_help_default(topic, par);
    }
@@ -1639,7 +1804,7 @@ int read_yesno(void)
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
 {
    int ret;
-    FILE *f = fopen(filename, "rb");
+    FILE *f = av_fopen_utf8(filename, "rb");

    if (!f) {
        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
@@ -1816,13 +1981,13 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
 {
    if (new_size >= INT_MAX / elem_size) {
        av_log(NULL, AV_LOG_ERROR, "Array too big.\n");
-        exit(1);
+        exit_program(1);
    }
    if (*size < new_size) {
        uint8_t *tmp = av_realloc(array, new_size*elem_size);
        if (!tmp) {
            av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
-            exit(1);
+            exit_program(1);
        }
        memset(tmp + *size*elem_size, 0, (new_size-*size) * elem_size);
        *size = new_size;
@@ -1830,144 +1995,3 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
    }
    return array;
 }
-
-static int alloc_buffer(FrameBuffer **pool, AVCodecContext *s, FrameBuffer **pbuf)
-{
-    const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(s->pix_fmt);
-    FrameBuffer *buf;
-    int i, ret;
-    int pixel_size;
-    int h_chroma_shift, v_chroma_shift;
-    int edge = 32; // XXX should be avcodec_get_edge_width(), but that fails on svq1
-    int w = s->width, h = s->height;
-
-    if (!desc)
-        return AVERROR(EINVAL);
-    pixel_size = desc->comp[0].step_minus1 + 1;
-
-    buf = av_mallocz(sizeof(*buf));
-    if (!buf)
-        return AVERROR(ENOMEM);
-
-    avcodec_align_dimensions(s, &w, &h);
-
-    if (!(s->flags & CODEC_FLAG_EMU_EDGE)) {
-        w += 2*edge;
-        h += 2*edge;
-    }
-
-    if ((ret = av_image_alloc(buf->base, buf->linesize, w, h,
-                              s->pix_fmt, 32)) < 0) {
-        av_freep(&buf);
-        av_log(s, AV_LOG_ERROR, "alloc_buffer: av_image_alloc() failed\n");
-        return ret;
-    }
-
-    avcodec_get_chroma_sub_sample(s->pix_fmt, &h_chroma_shift, &v_chroma_shift);
-    for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
-        const int h_shift = i==0 ? 0 : h_chroma_shift;
-        const int v_shift = i==0 ? 0 : v_chroma_shift;
-        if ((s->flags & CODEC_FLAG_EMU_EDGE) || !buf->linesize[i] || !buf->base[i])
-            buf->data[i] = buf->base[i];
-        else
-            buf->data[i] = buf->base[i] +
-                           FFALIGN((buf->linesize[i]*edge >> v_shift) +
-                                   (pixel_size*edge >> h_shift), 32);
-    }
-    buf->w       = s->width;
-    buf->h       = s->height;
-    buf->pix_fmt = s->pix_fmt;
-    buf->pool    = pool;
-
-    *pbuf = buf;
-    return 0;
-}
-
-int codec_get_buffer(AVCodecContext *s, AVFrame *frame)
-{
-    FrameBuffer **pool = s->opaque;
-    FrameBuffer *buf;
-    int ret, i;
-
-    if(av_image_check_size(s->width, s->height, 0, s) || s->pix_fmt<0) {
-        av_log(s, AV_LOG_ERROR, "codec_get_buffer: image parameters invalid\n");
-        return -1;
-    }
-
-    if (!*pool && (ret = alloc_buffer(pool, s, pool)) < 0)
-        return ret;
-
-    buf              = *pool;
-    *pool            = buf->next;
-    buf->next        = NULL;
-    if (buf->w != s->width || buf->h != s->height || buf->pix_fmt != s->pix_fmt) {
-        av_freep(&buf->base[0]);
-        av_free(buf);
-        if ((ret = alloc_buffer(pool, s, &buf)) < 0)
-            return ret;
-    }
-    av_assert0(!buf->refcount);
-    buf->refcount++;
-
-    frame->opaque        = buf;
-    frame->type          = FF_BUFFER_TYPE_USER;
-    frame->extended_data = frame->data;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
-        frame->base[i]     = buf->base[i];  // XXX h264.c uses base though it shouldn't
-        frame->data[i]     = buf->data[i];
-        frame->linesize[i] = buf->linesize[i];
-    }
-
-    return 0;
-}
-
-static void unref_buffer(FrameBuffer *buf)
-{
-    FrameBuffer **pool = buf->pool;
-
-    av_assert0(buf->refcount > 0);
-    buf->refcount--;
-    if (!buf->refcount) {
-        FrameBuffer *tmp;
-        for(tmp= *pool; tmp; tmp= tmp->next)
-            av_assert1(tmp != buf);
-
-        buf->next = *pool;
-        *pool = buf;
-    }
-}
-
-void codec_release_buffer(AVCodecContext *s, AVFrame *frame)
-{
-    FrameBuffer *buf = frame->opaque;
-    int i;
-
-    if(frame->type!=FF_BUFFER_TYPE_USER) {
-        avcodec_default_release_buffer(s, frame);
-        return;
-    }
-
-    for (i = 0; i < FF_ARRAY_ELEMS(frame->data); i++)
-        frame->data[i] = NULL;
-
-    unref_buffer(buf);
-}
-
-void filter_release_buffer(AVFilterBuffer *fb)
-{
-    FrameBuffer *buf = fb->priv;
-    av_free(fb);
-    unref_buffer(buf);
-}
-
-void free_buffer_pool(FrameBuffer **pool)
-{
-    FrameBuffer *buf = *pool;
-    while (buf) {
-        *pool = buf->next;
-        av_freep(&buf->base[0]);
-        av_free(buf);
-        buf = *pool;
-    }
-}
@@ -43,16 +43,22 @@ extern const char program_name[];
 */
 extern const int program_birth_year;

-/**
- * this year, defined by the program for show_banner()
- */
-extern const int this_year;
-
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
+extern int hide_banner;
+
+/**
+ * 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
@@ -71,6 +77,11 @@ void uninit_opts(void);
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);

+/**
+ * Override the cpuflags.
+ */
+int opt_cpuflags(void *optctx, const char *opt, const char *arg);
+
 /**
 * Fallback for options that are not explicitly handled, these will be
 * parsed through AVOptions.
@@ -86,10 +97,14 @@ int opt_report(const char *opt);

 int opt_max_alloc(void *optctx, const char *opt, const char *arg);

-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
-
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);

+#if CONFIG_OPENCL
+int opt_opencl(void *optctx, const char *opt, const char *arg);
+
+int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
+#endif
+
 /**
 * Limit the execution time.
 */
@@ -162,6 +177,8 @@ typedef struct OptionDef {
                                   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 +207,13 @@ 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);

@@ -242,6 +259,11 @@ typedef struct OptionGroupDef {
     * 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 {
@@ -392,6 +414,13 @@ void show_banner(int argc, char **argv, const OptionDef *options);
 */
 int show_version(void *optctx, const char *opt, const char *arg);

+/**
+ * Print the build configuration of the program to stdout. The contents
+ * depend on the definition of FFMPEG_CONFIGURATION.
+ * This option processing function does not utilize the arguments.
+ */
+int show_buildconf(void *optctx, const char *opt, const char *arg);
+
 /**
 * Print the license of the program to stdout. The license depends on
 * the license of the libraries compiled into the program.
@@ -466,6 +495,12 @@ int show_layouts(void *optctx, const char *opt, const char *arg);
 */
 int show_sample_fmts(void *optctx, const char *opt, const char *arg);

+/**
+ * Print a listing containing all the color names and values recognized
+ * by the program.
+ */
+int show_colors(void *optctx, const char *opt, const char *arg);
+
 /**
 * Return a positive value if a line read from standard input
 * starts with [yY], otherwise return 0.
@@ -479,7 +514,7 @@ int read_yesno(void);
 * @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
+ * @return >= 0 in case of success, a negative value corresponding to an
 * AVERROR error code in case of failure.
 */
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
@@ -517,52 +552,11 @@ FILE *get_preset_file(char *filename, size_t filename_size,
 */
 void *grow_array(void *array, int elem_size, int *size, int new_size);

+#define media_type_string av_get_media_type_string
+
 #define GROW_ARRAY(array, nb_elems)\
    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)

-typedef struct FrameBuffer {
-    uint8_t *base[4];
-    uint8_t *data[4];
-    int  linesize[4];
-
-    int h, w;
-    enum AVPixelFormat pix_fmt;
-
-    int refcount;
-    struct FrameBuffer **pool;  ///< head of the buffer pool
-    struct FrameBuffer *next;
-} FrameBuffer;
-
-/**
- * Get a frame from the pool. This is intended to be used as a callback for
- * AVCodecContext.get_buffer.
- *
- * @param s codec context. s->opaque must be a pointer to the head of the
- *          buffer pool.
- * @param frame frame->opaque will be set to point to the FrameBuffer
- *              containing the frame data.
- */
-int codec_get_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVCodecContext.release_buffer along with
- * codec_get_buffer().
- */
-void codec_release_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVFilterBuffer.free.
- * @param fb buffer to free. fb->priv must be a pointer to the FrameBuffer
- *           containing the buffer data.
- */
-void filter_release_buffer(AVFilterBuffer *fb);
-
-/**
- * Free all the buffers in the pool. This must be called after all the
- * buffers have been released.
- */
-void free_buffer_pool(FrameBuffer **pool);
-
 #define GET_PIX_FMT_NAME(pix_fmt)\
    const char *name = av_get_pix_fmt_name(pix_fmt);

@@ -4,6 +4,7 @@
    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
+    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
@@ -14,8 +15,14 @@
    { "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" },
+    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
+    { "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" },
+    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
+    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
+#if CONFIG_OPENCL
+    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
+    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
+#endif
@@ -0,0 +1,274 @@
+/*
+ * Copyright (C) 2013 Lenny Wang
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "libavutil/log.h"
+#include "libavutil/opencl.h"
+#include "cmdutils.h"
+
+typedef struct {
+    int platform_idx;
+    int device_idx;
+    char device_name[64];
+    int64_t runtime;
+} OpenCLDeviceBenchmark;
+
+const char *ocl_bench_source = AV_OPENCL_KERNEL(
+inline unsigned char clip_uint8(int a)
+{
+    if (a & (~0xFF))
+        return (-a)>>31;
+    else
+        return a;
+}
+
+kernel void unsharp_bench(
+                    global unsigned char *src,
+                    global unsigned char *dst,
+                    global int *mask,
+                    int width,
+                    int height)
+{
+    int i, j, local_idx, lc_idx, sum = 0;
+    int2 thread_idx, block_idx, global_idx, lm_idx;
+    thread_idx.x = get_local_id(0);
+    thread_idx.y = get_local_id(1);
+    block_idx.x = get_group_id(0);
+    block_idx.y = get_group_id(1);
+    global_idx.x = get_global_id(0);
+    global_idx.y = get_global_id(1);
+    local uchar data[32][32];
+    local int lc[128];
+
+    for (i = 0; i <= 1; i++) {
+        lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
+        lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
+        lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
+        for (j = 0; j <= 1; j++) {
+            lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
+            lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
+            lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
+            data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
+        }
+    }
+    local_idx = thread_idx.y*16 + thread_idx.x;
+    if (local_idx < 128)
+        lc[local_idx] = mask[local_idx];
+    barrier(CLK_LOCAL_MEM_FENCE);
+
+    \n#pragma unroll\n
+    for (i = -4; i <= 4; i++) {
+        lm_idx.y = 8 + i + thread_idx.y;
+        \n#pragma unroll\n
+        for (j = -4; j <= 4; j++) {
+            lm_idx.x = 8 + j + thread_idx.x;
+            lc_idx = (i + 4)*8 + j + 4;
+            sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
+        }
+    }
+    int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
+    int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
+    if (global_idx.x < width && global_idx.y < height)
+        dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
+}
+);
+
+#define OCLCHECK(method, ... )                                                 \
+do {                                                                           \
+    status = method(__VA_ARGS__);                                              \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n",                   \
+               av_opencl_errstr(status));                                      \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+#define CREATEBUF(out, flags, size)                                            \
+do {                                                                           \
+    out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n");        \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+static void fill_rand_int(int *data, int n)
+{
+    int i;
+    srand(av_gettime());
+    for (i = 0; i < n; i++)
+        data[i] = rand();
+}
+
+#define OPENCL_NB_ITER 5
+static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
+{
+    int i, arg = 0, width = 1920, height = 1088;
+    int64_t start, ret = 0;
+    cl_int status;
+    size_t kernel_len;
+    char *inbuf;
+    int *mask;
+    int buf_size = width * height * sizeof(char);
+    int mask_size = sizeof(uint32_t) * 128;
+
+    cl_mem cl_mask, cl_inbuf, cl_outbuf;
+    cl_kernel kernel = NULL;
+    cl_program program = NULL;
+    size_t local_work_size_2d[2] = {16, 16};
+    size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
+
+    if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
+        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fill_rand_int((int*)inbuf, buf_size/4);
+    fill_rand_int(mask, mask_size/4);
+
+    CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
+    CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
+    CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
+
+    kernel_len = strlen(ocl_bench_source);
+    program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
+                                        &kernel_len, &status);
+    if (status != CL_SUCCESS || !program) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    kernel = clCreateKernel(program, "unsharp_bench", &status);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
+             buf_size, inbuf, 0, NULL, NULL);
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
+             mask_size, mask, 0, NULL, NULL);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
+
+    start = av_gettime();
+    for (i = 0; i < OPENCL_NB_ITER; i++)
+        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
+                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
+    clFinish(ext_opencl_env->command_queue);
+    ret = (av_gettime() - start)/OPENCL_NB_ITER;
+end:
+    if (kernel)
+        clReleaseKernel(kernel);
+    if (program)
+        clReleaseProgram(program);
+    if (cl_inbuf)
+        clReleaseMemObject(cl_inbuf);
+    if (cl_outbuf)
+        clReleaseMemObject(cl_outbuf);
+    if (cl_mask)
+        clReleaseMemObject(cl_mask);
+    av_free(inbuf);
+    av_free(mask);
+    return ret;
+}
+
+static int compare_ocl_device_desc(const void *a, const void *b)
+{
+    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
+}
+
+int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
+{
+    int i, j, nb_devices = 0, count = 0;
+    int64_t score = 0;
+    AVOpenCLDeviceList *device_list;
+    AVOpenCLDeviceNode *device_node = NULL;
+    OpenCLDeviceBenchmark *devices = NULL;
+    cl_platform_id platform;
+
+    av_opencl_get_device_list(&device_list);
+    for (i = 0; i < device_list->platform_num; i++)
+        nb_devices += device_list->platform_node[i]->device_num;
+    if (!nb_devices) {
+        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
+        return AVERROR(EINVAL);
+    }
+    if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
+        return AVERROR(ENOMEM);
+    }
+
+    for (i = 0; i < device_list->platform_num; i++) {
+        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
+            device_node = device_list->platform_node[i]->device_node[j];
+            platform = device_list->platform_node[i]->platform_id;
+            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
+            if (score > 0) {
+                devices[count].platform_idx = i;
+                devices[count].device_idx = j;
+                devices[count].runtime = score;
+                strcpy(devices[count].device_name, device_node->device_name);
+                count++;
+            }
+        }
+    }
+    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
+    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
+    for (i = 0; i < count; i++)
+        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
+                devices[i].platform_idx, devices[i].device_idx,
+                devices[i].device_name, devices[i].runtime);
+
+    av_opencl_free_device_list(&device_list);
+    av_free(devices);
+    return 0;
+}
+
+int opt_opencl(void *optctx, const char *opt, const char *arg)
+{
+    char *key, *value;
+    const char *opts = arg;
+    int ret = 0;
+    while (*opts) {
+        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
+        if (ret < 0)
+            return ret;
+        ret = av_opencl_set_option(key, value);
+        if (ret < 0)
+            return ret;
+        if (*opts)
+            opts++;
+    }
+    return ret;
+}
@@ -10,7 +10,7 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM

 MSG    = $@
@@ -43,6 +43,7 @@ endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
+COMPILE_HOSTC = $(call COMPILE,HOSTCC)

 %.o: %.c
 	$(COMPILE_C)
@@ -56,6 +57,12 @@ COMPILE_S = $(call COMPILE,AS)
 %.o: %.S
 	$(COMPILE_S)

+%_host.o: %.c
+	$(COMPILE_HOSTC)
+
+%.o: %.rc
+	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
+
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<

@@ -82,14 +89,15 @@ endif
 include $(SRC_PATH)/arch.mak

 OBJS      += $(OBJS-yes)
+SLIBOBJS  += $(SLIBOBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)

 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)

-EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
+SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -99,8 +107,10 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)

-DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))

+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)%)
@@ -111,18 +121,19 @@ checkheaders: $(HOBJS)
 alltools: $(TOOLS)

 $(HOSTOBJS): %.o: %.c
-	$(call COMPILE,HOSTCC)
+	$(COMPILE_HOSTC)

 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)

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

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

 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
@@ -137,4 +148,4 @@ endef

 $(eval $(RULES))

-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
@@ -0,0 +1,31 @@
+/*
+ * Work around the class() function in AIX math.h clashing with
+ * identifiers named "class".
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef FFMPEG_COMPAT_AIX_MATH_H
+#define FFMPEG_COMPAT_AIX_MATH_H
+
+#define class class_in_math_h_causes_problems
+
+#include_next <math.h>
+
+#undef class
+
+#endif /* FFMPEG_COMPAT_AIX_MATH_H */
@@ -0,0 +1,880 @@
+// 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., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 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
@@ -0,0 +1,68 @@
+//  Copyright (c) 2011 FFmpegSource Project
+//
+//  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.
+
+/* these are defines/functions that are used and were changed in the switch to 2.6
+ * and are needed to maintain full compatility with 2.5 */
+
+enum {
+  AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
+  AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
+};
+
+AVSC_INLINE int avs_get_height_p_25(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 int avs_get_row_size_p_25(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_is_yv12_25(const AVS_VideoInfo * p)
+    { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }
@@ -0,0 +1,728 @@
+// 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., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 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__
@@ -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__
@@ -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__
@@ -38,8 +38,6 @@ static int optind = 1;
 static int optopt;
 static char *optarg;

-#undef fprintf
-
 static int getopt(int argc, char *argv[], char *opts)
 {
    static int sp = 1;
@@ -32,6 +32,8 @@
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>

+#include "libavutil/mem.h"
+
 typedef TID  pthread_t;
 typedef void pthread_attr_t;

@@ -1,3 +1,24 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef FFMPEG_COMPAT_TMS470_MATH_H
+#define FFMPEG_COMPAT_TMS470_MATH_H
+
 #include_next <math.h>

 #undef INFINITY
@@ -5,3 +26,5 @@

 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
+
+#endif /* FFMPEG_COMPAT_TMS470_MATH_H */
@@ -24,3 +24,6 @@
 #if !defined(va_copy) && defined(_MSC_VER)
 #define va_copy(dst, src) ((dst) = (src))
 #endif
+#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
+#define va_copy(dst, src) __va_copy(dst, src)
+#endif
@@ -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 FFMPEG_COMPAT_W32PTHREADS_H
+#define FFMPEG_COMPAT_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
@@ -62,11 +62,18 @@ typedef struct pthread_cond_t {
 } pthread_cond_t;

 /* function pointers to conditional variable API on windows 6.0+ kernels */
+#if _WIN32_WINNT < 0x0600
 static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
 static void (WINAPI *cond_init)(pthread_cond_t *cond);
 static void (WINAPI *cond_signal)(pthread_cond_t *cond);
 static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
                                DWORD milliseconds);
+#else
+#define cond_init      InitializeConditionVariable
+#define cond_broadcast WakeAllConditionVariable
+#define cond_signal    WakeConditionVariable
+#define cond_wait      SleepConditionVariableCS
+#endif

 static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
@@ -257,6 +264,7 @@ 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 +275,8 @@ static void w32thread_init(void)
        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
    cond_wait      =
        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
+#endif
+
 }

-#endif /* AVCODEC_W32PTHREADS_H */
+#endif /* FFMPEG_COMPAT_W32PTHREADS_H */
@@ -0,0 +1,132 @@
+#!/bin/sh
+
+# Copyright (c) 2013, Derek Buitenhuis
+#
+# Permission to use, copy, modify, and/or distribute this software for any
+# purpose with or without fee is hereby granted, provided that the above
+# copyright notice and this permission notice appear in all copies.
+#
+# THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
+# WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
+# MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
+# ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+# WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+# ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
+# OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
+
+# mktemp isn't POSIX, so supply an implementation
+mktemp() {
+    echo "${2%%XXX*}.${HOSTNAME}.${UID}.$$"
+}
+
+if [ $# -lt 2 ]; then
+    echo "Usage: makedef <version_script> <objects>" >&2
+    exit 0
+fi
+
+vscript=$1
+shift
+
+if [ ! -f "$vscript" ]; then
+    echo "Version script does not exist" >&2
+    exit 1
+fi
+
+for object in "$@"; do
+    if [ ! -f "$object" ]; then
+        echo "Object does not exist: ${object}" >&2
+        exit 1
+    fi
+done
+
+# Create a lib temporarily to dump symbols from.
+# It's just much easier to do it this way
+libname=$(mktemp -u "library").lib
+
+trap 'rm -f -- $libname' EXIT
+
+lib -out:${libname} $@ >/dev/null
+if [ $? != 0 ]; then
+    echo "Could not create temporary library." >&2
+    exit 1
+fi
+
+IFS='
+'
+
+# Determine if we're building for x86 or x86_64 and
+# set the symbol prefix accordingly.
+prefix=""
+arch=$(dumpbin -headers ${libname} |
+       tr '\t' ' ' |
+       grep '^ \+.\+machine \+(.\+)' |
+       head -1 |
+       sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
+
+if [ "${arch}" = "x86" ]; then
+    prefix="_"
+else
+    if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
+        echo "Unknown machine type." >&2
+        exit 1
+    fi
+fi
+
+started=0
+regex="none"
+
+for line in $(cat ${vscript} | tr '\t' ' '); do
+    # We only care about global symbols
+    echo "${line}" | grep -q '^ \+global:'
+    if [ $? = 0 ]; then
+        started=1
+        line=$(echo "${line}" | sed -e 's/^ \{1,\}global: *//')
+    else
+        echo "${line}" | grep -q '^ \+local:'
+        if [ $? = 0 ]; then
+            started=0
+        fi
+    fi
+
+    if [ ${started} = 0 ]; then
+        continue
+    fi
+
+    # Handle multiple symbols on one line
+    IFS=';'
+
+    # Work around stupid expansion to filenames
+    line=$(echo "${line}" | sed -e 's/\*/.\\+/g')
+    for exp in ${line}; do
+        # Remove leading and trailing whitespace
+        exp=$(echo "${exp}" | sed -e 's/^ *//' -e 's/ *$//')
+
+        if [ "${regex}" = "none" ]; then
+            regex="${exp}"
+        else
+            regex="${regex};${exp}"
+        fi
+    done
+
+    IFS='
+'
+done
+
+dump=$(dumpbin -linkermember:1 ${libname})
+
+rm ${libname}
+
+IFS=';'
+list=""
+for exp in ${regex}; do
+    list="${list}"'
+'$(echo "${dump}" |
+          sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
+          tail -n +2 |
+          cut -d' ' -f3 |
+          grep "^${exp}" |
+          sed -e 's/^/    /')
+done
+
+echo "EXPORTS"
+echo "${list}" | sort | uniq | tail -n +2
@@ -2,10 +2,10 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.

 The last version increases were:
-libavcodec:    2012-01-27
-libavdevice:   2011-04-18
-libavfilter:   2012-06-22
-libavformat:   2012-01-27
+libavcodec:    2013-03-xx
+libavdevice:   2013-03-xx
+libavfilter:   2013-12-xx
+libavformat:   2013-03-xx
 libavresample: 2012-10-05
 libpostproc:   2011-04-18
 libswresample: 2011-09-19
@@ -15,10 +15,284 @@ libavutil:     2012-10-22

 API changes, most recent first:

-2013-03-07 - xxxxxx - lavu 52.18.100 - avstring.h,bprint.h
+2014-03-18 - e9c8a9a - lsws 2.5.102
+  Make gray16 full-scale.
+
+2014-xx-xx - xxxxxxx - lavu 53.05.0 - frame.h
+  Add av_frame_copy() for copying the frame data.
+
+2014-02-xx - xxxxxxx - lswr 0.18.100 - swresample.h
+  Add swr_is_initialized() for checking whether a resample context is initialized.
+
+2014-02-xx - xxxxxxx - lavr 1.2.0 - avresample.h
+  Add avresample_is_open() for checking whether a resample context is open.
+
+2014-xx-xx - xxxxxxx - lavu 53.04.0  - opt.h
+  Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only)
+  for reading.
+
+2014-xx-xx - xxxxxxx - lavu 53.03.01 - opt.h
+  Deprecate unused AV_OPT_FLAG_METADATA.
+
+2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
+  Add avdevice_list_devices() and avdevice_free_list_devices()
+
+2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
+  Add avio_find_protocol_name() to find out the name of the protocol that would
+  be selected for a given URL.
+
+2014-02-xx - xxxxxxx - lavu 53.3.0 - frame.h
+  Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and
+  downmix_info.h API, which identify downmix-related metadata.
+
+2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h
+  Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame().
+
+2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h
+  Add AVFormatContext.max_interleave_delta for controlling amount of buffering
+  when interleaving.
+
+2014-02-02 - xxxxxxx - lavf 55.29.100 - avformat.h
+  Add output_ts_offset muxing option to AVFormatContext.
+
+2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h
+                       lavf 55.28.100 - avformat.h
+  Add avdevice_dev_to_app_control_message() function.
+
+2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h
+                       lavf 55.27.100 - avformat.h
+  Add avdevice_app_to_dev_control_message() function.
+
+2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h
+  Add AVFormatContext option metadata_header_padding to allow control over the
+  amount of padding added.
+
+2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h
+  Edges are not required anymore on video buffers allocated by get_buffer2()
+  (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate
+  CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width().
+
+2014-01-19 - xxxxxxx - lavf 55.25.100 - avformat.h
+  Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
+
+2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
+  Add av_make_q() function.
+
+2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
+  Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which
+  identifies AVMatrixEncoding data.
+
+2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h
+  Add values for various Dolby flags to the AVMatrixEncoding enum.
+
+2014-01-04 - xxxxxxx - lavu 52.60.100 - mathematics.h
+  Add av_add_stable() function.
+
+2013-12-22 - xxxxxxx - lavu 52.59.100 - avstring.h
+  Add av_strnlen() function.
+
+2013-12-xx - xxxxxxx - lavu 52.57.100 - opencl.h
+  Add av_opencl_benchmark() function.
+
+2013-11-xx - xxxxxxx - lavu 52.56.100 - ffversion.h
+  Moves version.h to libavutil/ffversion.h.
+  Install ffversion.h and make it public.
+
+2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h
+  av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be
+  used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and
+  avcodec_free_frame() respectively. The latter three functions are deprecated.
+
+2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h
+  Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and
+  stereo3d.h API, that identify codec-independent stereo3d information.
+
+2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h
+  Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which
+  identifies ATSC A53 Part 4 Closed Captions data.
+
+2013-11-XX - xxxxxxx - lavu 52.54.100 - avstring.h
+  Add av_utf8_decode() function.
+
+2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h
+  Add HEVC profiles
+
+2013-11-xx - xxxxxxx - lavc 55.44.100 - avcodec.h
+  Add av_packet_{un,}pack_dictionary()
+  Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value
+  strings between a stream and the application.
+
+2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h
+  Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil.
+
+2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h
+  Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate'
+  private option of the mpegvideo encoder family.
+
+2013-11-14 - 31c09b7 / 728c465 - lavc 55.26.0 - vdpau.h
+  Add av_vdpau_get_profile().
+  Add av_vdpau_alloc_context(). This function must from now on be
+  used for allocating AVVDPAUContext.
+
+2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h
+                       lavu 52.51.100 - frame.h
+  Add ITU-R BT.2020 and other not yet included values to color primaries,
+  transfer characteristics and colorspaces.
+
+2013-11-04 - xxxxxxx - lavu 52.50.100 - avutil.h
+  Add av_fopen_utf8()
+
+2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
+  Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
+
+2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
+  Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
+
+2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
+  Add av_get_known_color_name().
+
+2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
+  Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
+  av_opt_get_channel_layout() and av_opt_set_channel_layout().
+
+2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
+  Change default scaler to bicubic
+
+2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
+  Add av_codec_get_max_lowres()
+
+2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
+  Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
+  decoders
+
+2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h
+  Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
+  implementations.
+
+2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h
+  Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
+  AV_PIX_FMT_NV20.
+
+2013-09-16 - c74c3fb / 3feb3d6 - lavu 52.44.100 / 52.15.0 - mem.h
+  Add av_reallocp.
+
+2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
+  avcodec_close() argument can be NULL.
+
+2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
+  avformat_close_input() argument can be NULL and point on NULL.
+
+2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
+  Add av_format_get_probe_score().
+
+2013-08-15 - 1e0e193 - lsws 2.5.100 -
+  Add a sws_dither AVOption, allowing to set the dither algorithm used
+
+2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
+  Add a render2 alternative to the render callback function.
+
+2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
+  Add allocation function for AVVDPAUContext, allowing
+  to extend it in the future without breaking ABI/API.
+
+2013-08-10 - 67a580f / 5a9a9d4 - lavc 55.25.100 / 55.16.0 - avcodec.h
+  Extend AVPacket API with av_packet_unref, av_packet_ref,
+  av_packet_move_ref, av_packet_copy_props, av_packet_free_side_data.
+
+2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
+  Deprecate the bitstream-related members from struct AVVDPAUContext.
+  The bistream buffers no longer need to be explicitly freed.
+
+2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
+  Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
+  and select the AV_PIX_FMT_VDPAU format with get_format() instead.
+
+2013-08-05 - 4ee0984 / a0ad5d0 - lavu 52.41.100 / 52.14.0 - pixfmt.h
+  Deprecate AV_PIX_FMT_VDPAU_*. Use AV_PIX_FMT_VDPAU instead.
+
+2013-08-02 - 82fdfe8 / a8b1927 - lavc 55.20.100 / 55.11.0 - avcodec.h
+  Add output_picture_number to AVCodecParserContext.
+
+2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h
+  Add avcodec_chroma_pos_to_enum()
+  Add avcodec_enum_to_chroma_pos()
+
+2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
+  Deprecate avfilter_graph_parse() in favor of the equivalent
+  avfilter_graph_parse_ptr().
+
+2013-06-24 - af5f9c0 / 95d5246 - lavc 55.17.100 / 55.10.0 - avcodec.h
+  Add MPEG-2 AAC profiles
+
+2013-06-25 - af5f9c0 / 95d5246 - lavf 55.10.100 - avformat.h
+  Add AV_DISPOSITION_* flags to indicate text track kind.
+
+2013-06-15 - 99b8cd0 - lavu 52.36.100
+  Add AVRIPEMD:
+   av_ripemd_alloc()
+   av_ripemd_init()
+   av_ripemd_update()
+   av_ripemd_final()
+
+2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
+  Add av_realloc_array and av_reallocp_array
+
+2013-05-30 - 682b227 - lavu 52.35.100
+  Add AVSHA512:
+   av_sha512_alloc()
+   av_sha512_init()
+   av_sha512_update()
+   av_sha512_final()
+
+2013-05-24 - 8d4e969 / 129bb23 - lavfi 3.10.0 / 3.70.100 - avfilter.h
+  Add support for slice multithreading to lavfi. Filters supporting threading
+  are marked with AVFILTER_FLAG_SLICE_THREADS.
+  New fields AVFilterContext.thread_type, AVFilterGraph.thread_type and
+  AVFilterGraph.nb_threads (accessible directly or through AVOptions) may be
+  used to configure multithreading.
+
+2013-05-24 - fe40a9f / 2a6eaea - lavu 52.12.0 / 52.34.100 - cpu.h
+  Add av_cpu_count() function for getting the number of logical CPUs.
+
+2013-05-24 - 0c25c39 / b493847 - lavc 55.7.0 / 55.12.100 - avcodec.h
+  Add picture_structure to AVCodecParserContext.
+
+2013-05-17 - 3a751ea - lavu 52.33.100 - opt.h
+  Add AV_OPT_TYPE_COLOR value to AVOptionType enum.
+
+2013-05-13 - e398416 - lavu 52.31.100 - mem.h
+  Add av_dynarray2_add().
+
+2013-05-12 - 1776177 - lavfi 3.65.100
+  Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags.
+
+2013-04-19 - 380cfce - lavc 55.4.100
+  Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec.
+
+2013-04-18 - 7c1a002 - lavf 55.3.100
+  The matroska demuxer can now output proper verbatim ASS packets. It will
+  become the default starting lavf 56.0.100.
+
+2013-04-10 - af0d270 - lavu 25.26.100 - avutil.h,opt.h
+  Add av_int_list_length()
+  and av_opt_set_int_list().
+
+2013-03-30 - 5c73645 - lavu 52.24.100 - samplefmt.h
+  Add av_samples_alloc_array_and_samples().
+
+2013-03-29 - ef7b6b4 - lavf 55.1.100 - avformat.h
+  Add av_guess_frame_rate()
+
+2013-03-20 - 8d928a9 - lavu 52.22.100 - opt.h
+  Add AV_OPT_TYPE_DURATION value to AVOptionType enum.
+
+2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
+  Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
+
+2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
  Add av_escape() and av_bprint_escape() API.

-2013-02-24 - xxxxxx - lavfi 3.41.100 - buffersink.h
+2013-02-24 - b59cd08 - lavfi 3.41.100 - buffersink.h
  Add sample_rates field to AVABufferSinkParams.

 2013-01-17 - a1a707f - lavf 54.61.100
@@ -31,7 +305,7 @@ API changes, most recent first:
  Add AVFilterLink.channels, avfilter_link_get_channels()
  and avfilter_ref_get_channels().

-2012-12-15 - 2ada584d - lavc 54.80.100 - avcodec.h
+2012-12-15 - 96d815fc - lavc 54.80.100 - avcodec.h
  Add pkt_size field to AVFrame.

 2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h
@@ -144,21 +418,93 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
  Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.

-2013-xx-xx - xxxxxxx - lavfi 3.4.0 - avfiltergraph.h
+2013-05-15 - ff46809 / e6c4ac7 - lavu 52.32.100 / 52.11.0 - pixdesc.h
+  Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*.
+
+2013-04-03 - 6fc58a8 / 507b1e4 - lavc 55.7.100 / 55.4.0 - avcodec.h
+  Add field_order to AVCodecParserContext.
+
+2013-04-19 - f4b05cd / 5e83d9a - lavc 55.5.100 / 55.2.0 - avcodec.h
+  Add CODEC_FLAG_UNALIGNED to allow decoders to produce unaligned output.
+
+2013-04-11 - lavfi 3.53.100 / 3.8.0
+  231fd44 / 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate
+            avfilterhraph.h, user applications should include just avfilter.h
+  86070b8 / bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and
+            avfilter_graph_add_filter().
+  4fde705 / 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the
+            filter.
+  710b0aa / 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter().
+  46de9ba / 1ba95a9 - Add avfilter_init_dict().
+  16fc24b / 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags.
+  f4db6bf / 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs.
+  835cc0f / fa2a34c - Add avfilter_next(), deprecate av_filter_next().
+            Deprecate avfilter_uninit().
+
+2013-04-09 - lavfi 3.51.100 / 3.7.0 - avfilter.h
+  0594ef0 / b439c99 - Add AVFilter.priv_class for exporting filter options through the
+            AVOptions API in the similar way private options work in lavc and lavf.
+  44d4488 / 8114c10 - Add avfilter_get_class().
+  Switch all filters to use AVOptions.
+
+2013-03-19 - 17ebef2 / 2c328a9 - lavu 52.20.100 / 52.9.0 - pixdesc.h
+  Add av_pix_fmt_count_planes() function for counting planes in a pixel format.
+
+2013-03-16 - ecade98 / 42c7c61 - lavfi 3.47.100 / 3.6.0
+  Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count.
+
+2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.100 / 55.0.0, lavf 55.0.100 / 55.0.0,
+lavd 54.4.100 / 54.0.0, lavfi 3.5.0
+  36099df / 8e401db, 532f31a / 1cec062 - add a new API for reference counted buffers and buffer
+                     pools (new header libavutil/buffer.h).
+  2653e12 / 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data.
+            Add av_packet_from_data() function for constructing packets from
+            av_malloc()ed data.
+  c4e8821 / 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add
+            AVFrame.buf/extended_buf to allow reference counting for the AVFrame
+            data. Add new API for working with reference-counted AVFrames.
+  80e9e63 / 759001c - add the refcounted_frames field to AVCodecContext to make audio and
+            video decoders return reference-counted frames. Add get_buffer2()
+            callback to AVCodecContext which allocates reference-counted frames.
+            Add avcodec_default_get_buffer2() as the default get_buffer2()
+            implementation.
+            Deprecate AVCodecContext.get_buffer() / release_buffer() /
+            reget_buffer(), avcodec_default_get_buffer(),
+            avcodec_default_reget_buffer(), avcodec_default_release_buffer().
+            Remove avcodec_default_free_buffers(), which should not have ever
+            been called from outside of lavc.
+            Deprecate the following AVFrame fields:
+                * base -- is now stored in AVBufferRef
+                * reference, type, buffer_hints -- are unnecessary in the new API
+                * hwaccel_picture_private, owner, thread_opaque -- should not
+                  have been acessed from outside of lavc
+                * qscale_table, qstride, qscale_type, mbskip_table, motion_val,
+                  mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
+                  which are not exported anymore.
+  a05a44e / 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add
+            av_buffersrc_add_frame(), deprecate av_buffersrc_buffer().
+            Add av_buffersink_get_frame() and av_buffersink_get_samples(),
+            deprecate av_buffersink_read() and av_buffersink_read_samples().
+            Deprecate AVFilterBufferRef and all functions for working with it.
+
+2013-03-17 - 6c17ff8 / 12c5c1d - lavu 52.19.100 / 52.8.0 - avstring.h
+  Add av_isdigit, av_isgraph, av_isspace, av_isxdigit.
+
+2013-02-23 - 71cf094 / 9f12235 - lavfi 3.40.100 / 3.4.0 - avfiltergraph.h
  Add resample_lavr_opts to AVFilterGraph for setting libavresample options
  for auto-inserted resample filters.

-2013-xx-xx - xxxxxxx - lavu 52.7.0 - dict.h
+2013-01-25 - e7e14bc / 38c1466 - lavu 52.17.100 / 52.7.0 - dict.h
  Add av_dict_parse_string() to set multiple key/value pairs at once from a
  string.

-2013-01-xx - xxxxxxx - lavu 52.6.0 - avstring.h
+2013-01-25 - 25be630 / b85a5e8 - lavu 52.16.100 / 52.6.0 - avstring.h
  Add av_strnstr()

-2013-01-xx - xxxxxxx - lavu 52.5.0 - hmac.h
+2013-01-15 - e7e0186 / 8ee288d - lavu 52.15.100 / 52.5.0 - hmac.h
  Add AVHMAC.

-2013-01-13 - xxxxxxx - lavc 54.87.100 / 54.36.0 - vdpau.h
+2013-01-13 - 8ee7b38 / 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h
  Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding.

 2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h
@@ -246,7 +592,7 @@ API changes, most recent first:
 2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h
  Add AVFMT_FLAG_NOBUFFER for low latency use cases.

-2012-07-10 - 5fade8a - lavu 51.37.0
+2012-07-10 - fbe0245 / f3e5e6f - lavu 51.65.100 / 51.37.0
  Add av_malloc_array() and av_mallocz_array()

 2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0
@@ -31,9 +31,9 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 2.2.10

-# With the PROJECT_LOGO tag one can specify an logo or icon that is included
+# With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
 # pixels and the maximum width should not exceed 200 pixels. Doxygen will
 # copy the logo to the output directory.
@@ -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.
@@ -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
@@ -709,7 +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
@@ -793,13 +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            =

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

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

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

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

 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
@@ -6,7 +6,6 @@ LIBRARIES-$(CONFIG_AVFORMAT)   += libavformat
 LIBRARIES-$(CONFIG_AVDEVICE)   += libavdevice
 LIBRARIES-$(CONFIG_AVFILTER)   += libavfilter

-COMPONENTS-yes = $(PROGS-yes)
 COMPONENTS-$(CONFIG_AVUTIL)     += ffmpeg-utils
 COMPONENTS-$(CONFIG_SWSCALE)    += ffmpeg-scaler
 COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
@@ -15,9 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters

-MANPAGES    = $(COMPONENTS-yes:%=doc/%.1)    $(LIBRARIES-yes:%=doc/%.3)
-PODPAGES    = $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html)     \
+MANPAGES1   = $(AVPROGS-yes:%=doc/%.1)    $(AVPROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
+MANPAGES    = $(MANPAGES1) $(MANPAGES3)
+PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
@@ -35,6 +36,25 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)

+DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
+DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
+DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
+DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
+DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
+DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
+DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
+DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
+DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
+ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
+
+DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+PROGS              += $(DOC_EXAMPLES)
+
 all-$(CONFIG_DOC): doc

 doc: documentation
@@ -42,7 +62,9 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)

-TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
+examples: $(DOC_EXAMPLES)
+
+TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)

 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
@@ -59,12 +81,22 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(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 $(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/%-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)
@@ -73,29 +105,61 @@ doc/%.3: doc/%.pod $(GENTEXI)
 	$(M)pod2man --section=3 --center=" " --release=" " $< > $@

 $(DOCS) doc/doxy/html: | doc/
+$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
+OBJDIRS += doc/examples

-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
+DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
+
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(DOXY_INPUT)
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXY_INPUT)
+
+install-doc: install-html install-man
+
+install-html:

 install-man:

+ifdef CONFIG_HTMLPAGES
+install-progs-$(CONFIG_DOC): install-html
+
+install-html: $(HTMLPAGES)
+	$(Q)mkdir -p "$(DOCDIR)"
+	$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
+endif
+
 ifdef CONFIG_MANPAGES
 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: uninstall-doc
+
+uninstall-doc: uninstall-html uninstall-man
+
+uninstall-html:
+	$(RM) -r "$(DOCDIR)"

 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
+	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))

 clean:: docclean

-docclean:
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
+distclean:: docclean
+	$(RM) doc/config.texi
+
+examplesclean:
+	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
+	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
+
+docclean: examplesclean
+	$(RM) $(CLEANSUFFIXES:%=doc/%)
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html

 -include $(wildcard $(DOCS:%=%.d))
@@ -1,7 +1,7 @@
 Release Notes
 =============

-* 0.10 "Freedom"  January, 2012
+* 2.2 "Muybridge"  March, 2014


 General notes
@@ -14,9 +14,3 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
-
-Of big interest to our Windows users, FFmpeg now supports building with the MSVC
-compiler. Since MSVC does not support C99 features used extensively by FFmpeg,
-this has been accomplished using a converter that turns C99 code to C89. See the
-platform-specific documentation for more detailed documentation on building
-FFmpeg with MSVC.
@@ -1,36 +0,0 @@
-AVUtil
-======
-libavutil is a small lightweight library of generally useful functions.
-It is not a library for code needed by both libavcodec and libavformat.
-
-
-Overview:
-=========
-adler32.c               adler32 checksum
-aes.c                   AES encryption and decryption
-fifo.c                  resizeable first in first out buffer
-intfloat_readwrite.c    portable reading and writing of floating point values
-log.c                   "printf" with context and level
-md5.c                   MD5 Message-Digest Algorithm
-rational.c              code to perform exact calculations with rational numbers
-tree.c                  generic AVL tree
-crc.c                   generic CRC checksumming code
-integer.c               128bit integer math
-lls.c
-mathematics.c           greatest common divisor, integer sqrt, integer log2, ...
-mem.c                   memory allocation routines with guaranteed alignment
-
-Headers:
-bswap.h                 big/little/native-endian conversion code
-x86_cpu.h               a few useful macros for unifying x86-64 and x86-32 code
-avutil.h
-common.h
-intreadwrite.h          reading and writing of unaligned big/little/native-endian integers
-
-
-Goals:
-======
-* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
-* Small (source and object)
-* Efficient (low CPU and memory usage)
-* Useful (avoid useless features almost no one needs)
@@ -17,9 +17,46 @@ 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

-@section dump_extradata
+Remove zero padding at the end of a packet.
+
+@section dump_extra
+
+Add extradata to the beginning of the filtered packets.
+
+The additional argument specifies which packets should be filtered.
+It accepts the values:
+@table @samp
+@item a
+add extradata to all key packets, but only if @var{local_header} is
+set in the @option{flags2} codec context field
+
+@item k
+add extradata to all key packets
+
+@item e
+add extradata to all packets
+@end table
+
+If not specified it is assumed @samp{k}.
+
+For example the following @command{ffmpeg} command forces a global
+header (thus disabling individual packet headers) in the H.264 packets
+generated by the @code{libx264} encoder, but corrects them by adding
+the header stored in extradata to the key packets:
+@example
+ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
+@end example

@section h264_mp4toannexb

@@ -80,12 +117,10 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

@section movsub

-@section mp3_header_compress
-
@section mp3_header_decompress

@section noise

-@section remove_extradata
+@section remove_extra

@c man end BITSTREAM FILTERS
@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
 with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.

-The option @code{-codecs} of the ff* tools will display the list of
+The option @code{-decoders} of the ff* tools will display the list of
 enabled decoders.

@c man end DECODERS
@@ -52,6 +52,37 @@ top-field-first is assumed
@chapter Audio Decoders
@c man begin AUDIO DECODERS

+A description of some of the currently available audio decoders
+follows.
+
+@section ac3
+
+AC-3 audio decoder.
+
+This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
+the undocumented RealAudio 3 (a.k.a. dnet).
+
+@subsection AC-3 Decoder Options
+
+@table @option
+
+@item -drc_scale @var{value}
+Dynamic Range Scale Factor. The factor to apply to dynamic range values
+from the AC-3 stream. This factor is applied exponentially.
+There are 3 notable scale factor ranges:
+@table @option
+@item drc_scale == 0
+DRC disabled. Produces full range audio.
+@item 0 < drc_scale <= 1
+DRC enabled.  Applies a fraction of the stream DRC value.
+Audio reproduction is between full range and full compression.
+@item drc_scale > 1
+DRC enabled. Applies drc_scale asymmetrically.
+Loud sounds are fully compressed.  Soft sounds are enhanced.
+@end table
+
+@end table
+
@section ffwavesynth

 Internal wave synthetizer.
@@ -60,6 +91,78 @@ 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
@@ -86,4 +189,45 @@ ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@end table

+@section libzvbi-teletext
+
+Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
+subtitles. Requires the presence of the libzvbi headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libzvbi}.
+
+@subsection Options
+
+@table @option
+@item txt_page
+List of teletext page numbers to decode. You may use the special * string to
+match all pages. Pages that do not match the specified list are dropped.
+Default value is *.
+@item txt_chop_top
+Discards the top teletext line. Default value is 1.
+@item txt_format
+Specifies the format of the decoded subtitles. The teletext decoder is capable
+of decoding the teletext pages to bitmaps or to simple text, you should use
+"bitmap" for teletext pages, because certain graphics and colors cannot be
+expressed in simple text. You might use "text" for teletext based subtitles if
+your application can handle simple text based subtitles. Default value is
+bitmap.
+@item txt_left
+X offset of generated bitmaps, default is 0.
+@item txt_top
+Y offset of generated bitmaps, default is 0.
+@item txt_chop_spaces
+Chops leading and trailing spaces and removes empty lines from the generated
+text. This option is useful for teletext based subtitles where empty spaces may
+be present at the start or at the end of the lines or empty lines may be
+present between the subtitle lines because of double-sized teletext charactes.
+Default value is 1.
+@item txt_duration
+Sets the display duration of the decoded teletext pages or subtitles in
+miliseconds. Default value is 30000 which is 30 seconds.
+@item txt_transparent
+Force transparent background of the generated teletext bitmaps. Default value
+is 0 which means an opaque (black) background.
+@end table
+
@c man end SUBTILES DECODERS
@@ -1,3 +1,7 @@
+a.summary-letter {
+    text-decoration: none;
+}
+
 a {
    color: #2D6198;
 }
@@ -13,8 +17,8 @@ a:visited {
 }

 #banner img {
-    padding-bottom: 1px;
-    padding-top: 5px;
+    margin-bottom: 1px;
+    margin-top: 5px;
 }

 #body {
@@ -45,11 +49,16 @@ body {
    text-align: center;
 }

-h1, h2, h3 {
+h1 a, h2 a, h3 a, h4 a {
+    text-decoration: inherit;
+    color: inherit;
+}
+
+h1, h2, h3, h4 {
    padding-left: 0.4em;
    border-radius: 4px;
-    padding-bottom: 0.2em;
-    padding-top: 0.2em;
+    padding-bottom: 0.25em;
+    padding-top: 0.25em;
    border: 1px solid #6A996A;
 }

@@ -63,15 +72,22 @@ h1 {

 h2 {
    color: #313131;
-    font-size: 0.9em;
+    font-size: 1.0em;
    background-color: #ABE3AB;
 }

 h3 {
+    color: #313131;
+    font-size: 0.9em;
+    margin-bottom: -6px;
+    background-color: #BBF3BB;
+}
+
+h4 {
    color: #313131;
    font-size: 0.8em;
    margin-bottom: -8px;
-    background-color: #BBF3BB;
+    background-color: #D1FDD1;
 }

 img {
@@ -1,7 +1,7 @@
@chapter Demuxers
@c man begin DEMUXERS

-Demuxers are configured elements in FFmpeg which allow to read the
+Demuxers are configured elements in FFmpeg that can read the
 multimedia streams from a particular type of file.

 When you configure your FFmpeg build, all the supported demuxers
@@ -29,6 +29,17 @@ the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".

+@section asf
+
+Advanced Systems Format demuxer.
+
+This demuxer is used to demux ASF files and MMS network streams.
+
+@table @option
+@item -no_resync_search @var{bool}
+Do not try to resynchronize by looking for a certain optional start code.
+@end table
+
@anchor{concat}
@section concat

@@ -78,6 +89,9 @@ 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
@@ -100,6 +114,42 @@ probed and 0 otherwise.

@end table

+@section flv
+
+Adobe Flash Video Format demuxer.
+
+This demuxer is used to demux FLV files and RTMP network streams.
+
+@table @option
+@item -flv_metadata @var{bool}
+Allocate the streams according to the onMetaData array content.
+@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.
@@ -117,7 +167,7 @@ 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
@@ -195,6 +245,10 @@ 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.
@@ -208,28 +262,41 @@ Use @command{ffmpeg} for creating a video from the images in the file
 sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
 input frame rate of 10 frames per second:
@example
-ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
+ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
@end example

@item
 As above, but start by reading from a file with index 100 in the sequence:
@example
-ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
+ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
@end example

@item
 Read images matching the "*.png" glob pattern , that is all the files
 terminating with the ".png" suffix:
@example
-ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
+ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize

+@section mpegts
+
+MPEG-2 transport stream demuxer.
+
+@table @option
+
+@item fix_teletext_pts
+Overrides teletext packet PTS and DTS values with the timestamps calculated
+from the PCR of the first program which the teletext stream is part of and is
+not discarded. Default value is 1, set this option to 0 if you want your
+teletext packet PTS and DTS values untouched.
+@end table
+
@section rawvideo

 Raw video demuxer.

-This demuxer allows to read raw video data. Since there is no header
+This demuxer allows one 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.

@@ -11,29 +11,23 @@

@chapter Developers Guide

-@section API
-@itemize @bullet
-@item libavcodec is the library containing the codecs (both encoding and
-decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
-it.
+@section Notes for external developers

-@item libavformat is the library containing the file format handling (mux and
-demux code for several formats). Look at @file{ffplay.c} to use it in a
-player. See @file{doc/examples/muxing.c} to use it to generate audio or video
-streams.
+This document is mostly useful for internal FFmpeg developers.
+External developers who need to use the API in their application should
+refer to the API doxygen documentation in the public headers, and
+check the examples in @file{doc/examples} and in the source code to
+see how the public API is employed.

-@end itemize
+You can use the FFmpeg libraries in your commercial program, but you
+are encouraged to @emph{publish any patch you make}. In this case the
+best way to proceed is to send your patches to the ffmpeg-devel
+mailing list following the guidelines illustrated in the remainder of
+this document.

-@section Integrating libavcodec or libavformat in your program
-
-You can integrate all the source code of the libraries to link them
-statically to avoid any version problem. All you need is to provide a
-'config.mak' and a 'config.h' in the parent directory. See the defines
-generated by ./configure to understand what is needed.
-
-You can use libavcodec or libavformat in your commercial program, but
-@emph{any patch you make must be published}. The best way to proceed is
-to send your patches to the FFmpeg mailing list.
+For more detailed legal information about the use of FFmpeg in
+external programs read the @file{LICENSE} file in the source tree and
+consult @url{http://ffmpeg.org/legal.html}.

@section Contributing

@@ -57,13 +51,16 @@ and should try to fix issues their commit causes.
@subsection Code formatting conventions

 There are the following guidelines regarding the indentation in files:
+
@itemize @bullet
@item
 Indent size is 4.
+
@item
 The TAB character is forbidden outside of Makefiles as is any
 form of trailing whitespace. Commits containing either will be
 rejected by the git repository.
+
@item
 You should try to limit your code lines to 80 characters; however, do so if
 and only if this improves readability.
@@ -95,7 +92,7 @@ for markup commands, i.e. use @code{@@param} and not @code{\param}.
 * more text ...
 * ...
 */
-typedef struct Foobar@{
+typedef struct Foobar @{
    int var1; /**< var1 description */
    int var2; ///< var2 description
    /** var3 description */
@@ -117,13 +114,17 @@ int myfunc(int my_parameter)

 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
+
@itemize @bullet
@item
 the @samp{inline} keyword;
+
@item
@samp{//} comments;
+
@item
 designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
+
@item
 compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
@@ -135,13 +136,17 @@ clarity and performance.
 All code must compile with recent versions of GCC and a number of other
 currently supported compilers. To ensure compatibility, please do not use
 additional C99 features or GCC extensions. Especially watch out for:
+
@itemize @bullet
@item
 mixing statements and declarations;
+
@item
@samp{long long} (use @samp{int64_t} instead);
+
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
+
@item
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@@ -153,17 +158,25 @@ All names should be composed with underscores (_), not CamelCase. For example,
 for example structs and enums; they should always be in the CamelCase

 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 prefix is required.
+For file-scope variables and functions declared as @code{static}, no prefix
+is required.
+
@item
-For variables and functions used internally by a library an @code{ff_}
-prefix should be used, e.g. @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
 Each library has its own prefix for public symbols, in addition to the
 commonly used @code{av_} (@code{avformat_} for libavformat,
@@ -183,10 +196,12 @@ 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,
 please use av_log() instead.
+
@item
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
@@ -229,131 +244,154 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:

@enumerate
@item
-   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
-   You must not commit code which breaks FFmpeg! (Meaning unfinished but
-   enabled code which breaks compilation or compiles but does not work or
-   breaks the regression tests)
-   You can commit unfinished stuff (for testing etc), but it must be disabled
-   (#ifdef etc) by default so it does not interfere with other developers'
-   work.
-@item
-   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
-   (portability, triggers compiler bugs, unusual environment etc) they will be
-   reported and eventually fixed.
-@item
-   Do not commit unrelated changes together, split them into self-contained
-   pieces. Also do not forget that if part B depends on part A, but A does not
-   depend on B, then A can and should be committed first and separate from B.
-   Keeping changes well split into self-contained parts makes reviewing and
-   understanding them on the commit log mailing list easier. This also helps
-   in case of debugging later on.
-   Also if you have doubts about splitting or not splitting, do not hesitate to
-   ask/discuss it on the developer mailing list.
-@item
-   Do not change behavior of the programs (renaming options etc) or public
-   API or ABI without first discussing it on the ffmpeg-devel mailing list.
-   Do not remove functionality from the code. Just improve!
+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://opensource.org/licenses/isc-license.txt, 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.
+If you add a new file, give it a proper license header. Do not copy and
+paste it from a random place, use an existing file as template.

-   Note: Redundant code can be removed.
@item
-   Do not commit changes to the build system (Makefiles, configure script)
-   which change behavior, defaults etc, without asking first. The same
-   applies to compiler warning fixes, trivial looking fixes and to code
-   maintained by other developers. We usually have a reason for doing things
-   the way we do. Send your changes as patches to the ffmpeg-devel mailing
-   list, and if the code maintainers say OK, you may commit. This does not
-   apply to files you wrote and/or maintain.
-@item
-   We refuse source indentation and other cosmetic changes if they are mixed
-   with functional changes, such commits will be rejected and removed. Every
-   developer has his own indentation style, you should not change it. Of course
-   if you (re)write something, you can use your own style, even though we would
-   prefer if the indentation throughout FFmpeg was consistent (Many projects
-   force a given indentation style - we do not.). If you really need to make
-   indentation changes (try to avoid this), separate them strictly from real
-   changes.
+You must not commit code which breaks FFmpeg! (Meaning unfinished but
+enabled code which breaks compilation or compiles but does not work or
+breaks the regression tests)
+You can commit unfinished stuff (for testing etc), but it must be disabled
+(#ifdef etc) by default so it does not interfere with other developers'
+work.

-   NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
-   then either do NOT change the indentation of the inner part within (do not
-   move it to the right)! or do so in a separate commit
@item
-   Always fill out the commit log message. Describe in a few lines what you
-   changed and why. You can refer to mailing list postings if you fix a
-   particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
-   Recommended format:
-   area changed: Short 1 line description
+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.

-   details describing what and why and giving references.
@item
-   Make sure the author of the commit is set correctly. (see git commit --author)
-   If you apply a patch, send an
-   answer to ffmpeg-devel (or wherever you got the patch from) saying that
-   you applied the patch.
+You do not have to over-test things. If it works for you, and you think it
+should work for others, then commit. If your code has problems
+(portability, triggers compiler bugs, unusual environment etc) they will be
+reported and eventually fixed.
+
@item
-   When applying patches that have been discussed (at length) on the mailing
-   list, reference the thread in the log message.
+Do not commit unrelated changes together, split them into self-contained
+pieces. Also do not forget that if part B depends on part A, but A does not
+depend on B, then A can and should be committed first and separate from B.
+Keeping changes well split into self-contained parts makes reviewing and
+understanding them on the commit log mailing list easier. This also helps
+in case of debugging later on.
+Also if you have doubts about splitting or not splitting, do not hesitate to
+ask/discuss it on the developer mailing list.
+
@item
-    Do NOT commit to code actively maintained by others without permission.
-    Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
-    timeframe (12h for build failures and security fixes, 3 days small changes,
-    1 week for big patches) then commit your patch if you think it is OK.
-    Also note, the maintainer can simply ask for more time to review!
+Do not change behavior of the programs (renaming options etc) or public
+API or ABI without first discussing it on the ffmpeg-devel mailing list.
+Do not remove functionality from the code. Just improve!
+
+Note: Redundant code can be removed.
+
@item
-    Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
-    are sent there and reviewed by all the other developers. Bugs and possible
-    improvements or general questions regarding commits are discussed there. We
-    expect you to react if problems with your code are uncovered.
+Do not commit changes to the build system (Makefiles, configure script)
+which change behavior, defaults etc, without asking first. The same
+applies to compiler warning fixes, trivial looking fixes and to code
+maintained by other developers. We usually have a reason for doing things
+the way we do. Send your changes as patches to the ffmpeg-devel mailing
+list, and if the code maintainers say OK, you may commit. This does not
+apply to files you wrote and/or maintain.
+
@item
-    Update the documentation if you change behavior or add features. If you are
-    unsure how best to do this, send a patch to ffmpeg-devel, the documentation
-    maintainer(s) will review and commit your stuff.
+We refuse source indentation and other cosmetic changes if they are mixed
+with functional changes, such commits will be rejected and removed. Every
+developer has his own indentation style, you should not change it. Of course
+if you (re)write something, you can use your own style, even though we would
+prefer if the indentation throughout FFmpeg was consistent (Many projects
+force a given indentation style - we do not.). If you really need to make
+indentation changes (try to avoid this), separate them strictly from real
+changes.
+
+NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
+then either do NOT change the indentation of the inner part within (do not
+move it to the right)! or do so in a separate commit
+
@item
-    Try to keep important discussions and requests (also) on the public
-    developer mailing list, so that all developers can benefit from them.
+Always fill out the commit log message. Describe in a few lines what you
+changed and why. You can refer to mailing list postings if you fix a
+particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
+Recommended format:
+area changed: Short 1 line description
+
+details describing what and why and giving references.
+
@item
-    Never write to unallocated memory, never write over the end of arrays,
-    always check values read from some untrusted source before using them
-    as array index or other risky things.
+Make sure the author of the commit is set correctly. (see git commit --author)
+If you apply a patch, send an
+answer to ffmpeg-devel (or wherever you got the patch from) saying that
+you applied the patch.
+
@item
-    Remember to check if you need to bump versions for the specific libav*
-    parts (libavutil, libavcodec, libavformat) you are changing. You need
-    to change the version integer.
-    Incrementing the first component means no backward compatibility to
-    previous versions (e.g. removal of a function from the public API).
-    Incrementing the second component means backward compatible change
-    (e.g. addition of a function to the public API or extension of an
-    existing data structure).
-    Incrementing the third component means a noteworthy binary compatible
-    change (e.g. encoder bug fix that matters for the decoder). The third
-    component always starts at 100 to distinguish FFmpeg from Libav.
+When applying patches that have been discussed (at length) on the mailing
+list, reference the thread in the log message.
+
@item
-    Compiler warnings indicate potential bugs or code with bad style. If a type of
-    warning always points to correct and clean code, that warning should
-    be disabled, not the code changed.
-    Thus the remaining warnings can either be bugs or correct code.
-    If it is a bug, the bug has to be fixed. If it is not, the code should
-    be changed to not generate a warning unless that causes a slowdown
-    or obfuscates the code.
+Do NOT commit to code actively maintained by others without permission.
+Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+timeframe (12h for build failures and security fixes, 3 days small changes,
+1 week for big patches) then commit your patch if you think it is OK.
+Also note, the maintainer can simply ask for more time to review!
+
@item
-    If you add a new file, give it a proper license header. Do not copy and
-    paste it from a random place, use an existing file as template.
+Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+are sent there and reviewed by all the other developers. Bugs and possible
+improvements or general questions regarding commits are discussed there. We
+expect you to react if problems with your code are uncovered.
+
+@item
+Update the documentation if you change behavior or add features. If you are
+unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+maintainer(s) will review and commit your stuff.
+
+@item
+Try to keep important discussions and requests (also) on the public
+developer mailing list, so that all developers can benefit from them.
+
+@item
+Never write to unallocated memory, never write over the end of arrays,
+always check values read from some untrusted source before using them
+as array index or other risky things.
+
+@item
+Remember to check if you need to bump versions for the specific libav*
+parts (libavutil, libavcodec, libavformat) you are changing. You need
+to change the version integer.
+Incrementing the first component means no backward compatibility to
+previous versions (e.g. removal of a function from the public API).
+Incrementing the second component means backward compatible change
+(e.g. addition of a function to the public API or extension of an
+existing data structure).
+Incrementing the third component means a noteworthy binary compatible
+change (e.g. encoder bug fix that matters for the decoder). 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
+be disabled, not the code changed.
+Thus the remaining warnings can either be bugs or correct code.
+If it is a bug, the bug has to be fixed. If it is not, the code should
+be changed to not generate a warning unless that causes a slowdown
+or obfuscates the code.
+
+@item
+Make sure that no parts of the codebase that you maintain are missing from the
+@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
+your name after it.
+If at some point you no longer want to maintain some code, then please help
+finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
@end enumerate

 We think our rules are not too hard. If you have comments, contact us.
@@ -408,40 +446,51 @@ send a reminder by email. Your patch should eventually be dealt with.

@enumerate
@item
-    Did you use av_cold for codec initialization and close functions?
+Did you use av_cold for codec initialization and close functions?
+
@item
-    Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
-    AVInputFormat/AVOutputFormat struct?
+Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+AVInputFormat/AVOutputFormat struct?
+
@item
-    Did you bump the minor version number (and reset the micro version
-    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
+Did you bump the minor version number (and reset the micro version
+number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
+
@item
-    Did you register it in @file{allcodecs.c} or @file{allformats.c}?
+Did you register it in @file{allcodecs.c} or @file{allformats.c}?
+
@item
-    Did you add the AVCodecID to @file{avcodec.h}?
-    When adding new codec IDs, also add an entry to the codec descriptor
-    list in @file{libavcodec/codec_desc.c}.
+Did you add the AVCodecID to @file{avcodec.h}?
+When adding new codec IDs, also add an entry to the codec descriptor
+list in @file{libavcodec/codec_desc.c}.
+
@item
-    If it has a FourCC, did you add it to @file{libavformat/riff.c},
-    even if it is only a decoder?
+If it has a FourCC, did you add it to @file{libavformat/riff.c},
+even if it is only a decoder?
+
@item
-    Did you add a rule to compile the appropriate files in the Makefile?
-    Remember to do this even if you're just adding a format to a file that is
-    already being compiled by some other rule, like a raw demuxer.
+Did you add a rule to compile the appropriate files in the Makefile?
+Remember to do this even if you're just adding a format to a file that is
+already being compiled by some other rule, like a raw demuxer.
+
@item
-    Did you add an entry to the table of supported formats or codecs in
-    @file{doc/general.texi}?
+Did you add an entry to the table of supported formats or codecs in
+@file{doc/general.texi}?
+
@item
-    Did you add an entry in the Changelog?
+Did you add an entry in the Changelog?
+
@item
-    If it depends on a parser or a library, did you add that dependency in
-    configure?
+If it depends on a parser or a library, did you add that dependency in
+configure?
+
@item
-    Did you @code{git add} the appropriate files before committing?
+Did you @code{git add} the appropriate files before committing?
+
@item
-    Did you make sure it compiles standalone, i.e. with
-    @code{configure --disable-everything --enable-decoder=foo}
-    (or @code{--enable-demuxer} or whatever your component is)?
+Did you make sure it compiles standalone, i.e. with
+@code{configure --disable-everything --enable-decoder=foo}
+(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate


@@ -449,82 +498,109 @@ send a reminder by email. Your patch should eventually be dealt with.

@enumerate
@item
-    Does @code{make fate} pass with the patch applied?
+Does @code{make fate} pass with the patch applied?
+
@item
-    Was the patch generated with git format-patch or send-email?
+Was the patch generated with git format-patch or send-email?
+
@item
-    Did you sign off your patch? (git commit -s)
-    See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
-    of sign off.
+Did you sign off your patch? (git commit -s)
+See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+of sign off.
+
@item
-    Did you provide a clear git commit log message?
+Did you provide a clear git commit log message?
+
@item
-    Is the patch against latest FFmpeg git master branch?
+Is the patch against latest FFmpeg git master branch?
+
@item
-    Are you subscribed to ffmpeg-devel?
-    (the list is subscribers only due to spam)
+Are you subscribed to ffmpeg-devel?
+(the list is subscribers only due to spam)
+
@item
-    Have you checked that the changes are minimal, so that the same cannot be
-    achieved with a smaller patch and/or simpler final code?
+Have you checked that the changes are minimal, so that the same cannot be
+achieved with a smaller patch and/or simpler final code?
+
@item
-    If the change is to speed critical code, did you benchmark it?
+If the change is to speed critical code, did you benchmark it?
+
@item
-    If you did any benchmarks, did you provide them in the mail?
+If you did any benchmarks, did you provide them in the mail?
+
@item
-    Have you checked that the patch does not introduce buffer overflows or
-    other security issues?
+Have you checked that the patch does not introduce buffer overflows or
+other security issues?
+
@item
-    Did you test your decoder or demuxer against damaged data? If no, see
-    tools/trasher, 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.
+Did you test your decoder or demuxer against damaged data? If no, see
+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?
+Does the patch not mix functional and cosmetic changes?
+
@item
-    Did you add tabs or trailing whitespace to the code? Both are forbidden.
+Did you add tabs or trailing whitespace to the code? Both are forbidden.
+
@item
-    Is the patch attached to the email you send?
+Is the patch attached to the email you send?
+
@item
-    Is the mime type of the patch correct? It should be text/x-diff or
-    text/x-patch or at least text/plain and not application/octet-stream.
+Is the mime type of the patch correct? It should be text/x-diff or
+text/x-patch or at least text/plain and not application/octet-stream.
+
@item
-    If the patch fixes a bug, did you provide a verbose analysis of the bug?
+If the patch fixes a bug, did you provide a verbose analysis of the bug?
+
@item
-    If the patch fixes a bug, did you provide enough information, including
-    a sample, so the bug can be reproduced and the fix can be verified?
-    Note please do not attach samples >100k to mails but rather provide a
-    URL, you can upload to ftp://upload.ffmpeg.org
+If the patch fixes a bug, did you provide enough information, including
+a sample, so the bug can be reproduced and the fix can be verified?
+Note please do not attach samples >100k to mails but rather provide a
+URL, you can upload to ftp://upload.ffmpeg.org
+
@item
-    Did you provide a verbose summary about what the patch does change?
+Did you provide a verbose summary about what the patch does change?
+
@item
-    Did you provide a verbose explanation why it changes things like it does?
+Did you provide a verbose explanation why it changes things like it does?
+
@item
-    Did you provide a verbose summary of the user visible advantages and
-    disadvantages if the patch is applied?
+Did you provide a verbose summary of the user visible advantages and
+disadvantages if the patch is applied?
+
@item
-    Did you provide an example so we can verify the new feature added by the
-    patch easily?
+Did you provide an example so we can verify the new feature added by the
+patch easily?
+
@item
-    If you added a new file, did you insert a license header? It should be
-    taken from FFmpeg, not randomly copied and pasted from somewhere else.
+If you added a new file, did you insert a license header? It should be
+taken from FFmpeg, not randomly copied and pasted from somewhere else.
+
@item
-    You should maintain alphabetical order in alphabetically ordered lists as
-    long as doing so does not break API/ABI compatibility.
+You should maintain alphabetical order in alphabetically ordered lists as
+long as doing so does not break API/ABI compatibility.
+
@item
-    Lines with similar content should be aligned vertically when doing so
-    improves readability.
+Lines with similar content should be aligned vertically when doing so
+improves readability.
+
@item
-    Consider to add a regression test for your code.
+Consider to add a regression test for your code.
+
@item
-    If you added YASM code please check that things still work with --disable-yasm
+If you added YASM code please check that things still work with --disable-yasm
+
@item
-    Make sure you check the return values of function and return appropriate
-    error codes. Especially memory allocation functions like @code{av_malloc()}
-    are notoriously left unchecked, which is a serious problem.
+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.
+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
@@ -577,4 +653,145 @@ 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
@@ -0,0 +1,25 @@
+@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
+
+@ifclear config-writeonly
+@include indevs.texi
+@end ifclear
+@ifclear config-readonly
+@include outdevs.texi
+@end ifclear
@@ -8,7 +8,5 @@ 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
+EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
 EOF
@@ -1,9 +0,0 @@
-
- <footer class="footer pagination-right">
-   <span class="label label-info">
-    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
-   </span>
- </footer>
-</div>
-</body>
-</html>
@@ -1,16 +0,0 @@
-<!DOCTYPE html>
-<html>
-<head>
-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
-<meta http-equiv="X-UA-Compatible" content="IE=9"/>
-<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
-<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
-<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
-<!--Header replace -->
-
-</head>
-
-<div class="container">
-
-<!--Header replace -->
-<div class="menu">
@@ -1,288 +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 abs(x)
-Compute absolute value of @var{x}.
-
-@item acos(x)
-Compute arccosine of @var{x}.
-
-@item asin(x)
-Compute arcsine of @var{x}.
-
-@item atan(x)
-Compute arctangent of @var{x}.
-
-@item ceil(expr)
-Round the value of expression @var{expr} upwards to the nearest
-integer. For example, "ceil(1.5)" is "2.0".
-
-@item cos(x)
-Compute cosine of @var{x}.
-
-@item cosh(x)
-Compute hyperbolic cosine of @var{x}.
-
-@item eq(x, y)
-Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
-
-@item exp(x)
-Compute exponential of @var{x} (with base @code{e}, the Euler's number).
-
-@item floor(expr)
-Round the value of expression @var{expr} downwards to the nearest
-integer. For example, "floor(-1.5)" is "-2.0".
-
-@item gauss(x)
-Compute Gauss function of @var{x}, corresponding to
-@code{exp(-x*x/2) / sqrt(2*PI)}.
-
-@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 gt(x, y)
-Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
-
-@item gte(x, y)
-Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
-
-@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 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 if(x, y, z)
-Evaluate @var{x}, and if the result is non-zero return the evaluation
-result of @var{y}, otherwise the evaluation result of @var{z}.
-
-@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 ifnot(x, y, z)
-Evaluate @var{x}, and if the result is zero return the evaluation
-result of @var{y}, otherwise the evaluation result of @var{z}.
-
-@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 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 log(x)
-Compute natural logarithm of @var{x}.
-
-@item lt(x, y)
-Return 1 if @var{x} is lesser 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 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 mod(x, y)
-Compute the remainder of division of @var{x} by @var{y}.
-
-@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 print(t)
-@item print(t, l)
-Print the value of expression @var{t} with loglevel @var{l}. If
-@var{l} is not specified then a default log level is used.
-Returns the value of the expression printed.
-
-Prints t with loglevel l
-
-@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 root(expr, max)
-Find an input value for which the function represented by @var{expr}
-with argument @var{ld(0)} is 0 in the interval 0..@var{max}.
-
-The expression in @var{expr} must denote a continuous function or the
-result is undefined.
-
-@var{ld(0)} is used to represent the function input value, which means
-that the given expression will be evaluated multiple times with
-various input values that the expression can access through
-@code{ld(0)}. When the expression evaluates to 0 then the
-corresponding input value will be returned.
-
-@item sin(x)
-Compute sine of @var{x}.
-
-@item sinh(x)
-Compute hyperbolic sine of @var{x}.
-
-@item sqrt(expr)
-Compute the square root of @var{expr}. This is equivalent to
-"(@var{expr})^.5".
-
-@item squish(x)
-Compute expression @code{1/(1 + exp(4*x))}.
-
-@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 tan(x)
-Compute tangent of @var{x}.
-
-@item tanh(x)
-Compute hyperbolic tangent of @var{x}.
-
-@item taylor(expr, x)
-@item taylor(expr, x, id)
-Evaluate a Taylor series at @var{x}, given an expression representing
-the @code{ld(id)}-th derivative of a function at 0.
-
-When the series does not converge the result is undefined.
-
-@var{ld(id)} is used to represent the derivative order in @var{expr},
-which means that the given expression will be evaluated multiple times
-with various input values that the expression can access through
-@code{ld(id)}. If @var{id} is not specified then 0 is assumed.
-
-Note, when you have the derivatives at y instead of 0,
-@code{taylor(expr, x-y)} can be used.
-
-@item time(0)
-Return the current (wallclock) time in seconds.
-
-@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 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.
-@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
-
-For example the construct:
-@example
-if (A AND B) then C
-@end example
-is equivalent to:
-@example
-if(A*B, 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 unit prefixes.
-If 'i' is appended after the prefix, binary prefixes are used, which
-are based on powers of 1024 instead of powers of 1000.
-The 'B' postfix multiplies the value by 8, and can be appended after a
-unit prefix or used alone. This allows using for example 'KB', 'MiB',
-'G' and 'B' as number postfix.
-
-The list of available International System prefixes follows, with
-indication of the corresponding powers of 10 and of 2.
-@table @option
-@item y
-10^-24 / 2^-80
-@item z
-10^-21 / 2^-70
-@item a
-10^-18 / 2^-60
-@item f
-10^-15 / 2^-50
-@item p
-10^-12 / 2^-40
-@item n
-10^-9 / 2^-30
-@item u
-10^-6 / 2^-20
-@item m
-10^-3 / 2^-10
-@item c
-10^-2
-@item d
-10^-1
-@item h
-10^2
-@item k
-10^3 / 2^10
-@item K
-10^3 / 2^10
-@item M
-10^6 / 2^20
-@item G
-10^9 / 2^30
-@item T
-10^12 / 2^40
-@item P
-10^15 / 2^40
-@item E
-10^18 / 2^50
-@item Z
-10^21 / 2^60
-@item Y
-10^24 / 2^70
-@end table
-
-@c man end
@@ -7,24 +7,28 @@ 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)

-EXAMPLES=       decoding_encoding                  \
-                demuxing                           \
+EXAMPLES=       avio_reading                       \
+                avcodec                            \
+                demuxing_decoding                  \
                filtering_video                    \
                filtering_audio                    \
                metadata                           \
                muxing                             \
+                remuxing                           \
                resampling_audio                   \
                scaling_video                      \
+                transcode_aac                      \

 OBJS=$(addsuffix .o,$(EXAMPLES))

 # the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
+avcodec:           LDLIBS += -lm
 muxing:            LDLIBS += -lm
+resampling_audio:  LDLIBS += -lm

 .phony: all clean-test clean

@@ -5,14 +5,19 @@ 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
+Method 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
+Method 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.
+FFmpeg (no need to make install in any prefix). Then just run "make examples".
+This will build the examples using the FFmpeg build system. You can clean those
+examples using "make examplesclean"
+
+If you want to try the dedicated Makefile examples (to emulate the first
+method), go into doc/examples and run a command such as
+PKG_CONFIG_PATH=pc-uninstalled make.
@@ -24,10 +24,10 @@
 * @file
 * libavcodec API use example.
 *
+ * @example avcodec.c
 * 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>
@@ -79,7 +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;
@@ -88,9 +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++;
    }
@@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
    }

    /* frame containing input raw audio */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
@@ -170,6 +170,10 @@ static void audio_encode_example(const char *filename)
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
+    if (buffer_size < 0) {
+        fprintf(stderr, "Could not get sample buffer size\n");
+        exit(1);
+    }
    samples = av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
@@ -187,7 +191,7 @@ static void audio_encode_example(const char *filename)
    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for(i=0;i<200;i++) {
+    for (i = 0; i < 200; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;
@@ -227,7 +231,7 @@ static void audio_encode_example(const char *filename)
    fclose(f);

    av_freep(&samples);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
    avcodec_close(c);
    av_free(c);
 }
@@ -287,12 +291,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
        int got_frame = 0;

        if (!decoded_frame) {
-            if (!(decoded_frame = avcodec_alloc_frame())) {
+            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
-        } else
-            avcodec_get_frame_defaults(decoded_frame);
+        }

        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
@@ -304,6 +307,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
            int data_size = av_samples_get_buffer_size(NULL, c->channels,
                                                       decoded_frame->nb_samples,
                                                       c->sample_fmt, 1);
+            if (data_size < 0) {
+                /* This should not occur, checking just for paranoia */
+                fprintf(stderr, "Failed to calculate data size\n");
+                exit(1);
+            }
            fwrite(decoded_frame->data[0], 1, data_size, outfile);
        }
        avpkt.size -= len;
@@ -329,7 +337,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)

    avcodec_close(c);
    av_free(c);
-    avcodec_free_frame(&decoded_frame);
+    av_frame_free(&decoded_frame);
 }

 /*
@@ -366,12 +374,12 @@ static void video_encode_example(const char *filename, int codec_id)
    c->width = 352;
    c->height = 288;
    /* frames per second */
-    c->time_base= (AVRational){1,25};
+    c->time_base = (AVRational){1,25};
    c->gop_size = 10; /* emit one intra frame every ten frames */
-    c->max_b_frames=1;
+    c->max_b_frames = 1;
    c->pix_fmt = AV_PIX_FMT_YUV420P;

-    if(codec_id == AV_CODEC_ID_H264)
+    if (codec_id == AV_CODEC_ID_H264)
        av_opt_set(c->priv_data, "preset", "slow", 0);

    /* open it */
@@ -386,7 +394,7 @@ static void video_encode_example(const char *filename, int codec_id)
        exit(1);
    }

-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
@@ -405,7 +413,7 @@ static void video_encode_example(const char *filename, int codec_id)
    }

    /* encode 1 second of video */
-    for(i=0;i<25;i++) {
+    for (i = 0; i < 25; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL;    // packet data will be allocated by the encoder
        pkt.size = 0;
@@ -413,15 +421,15 @@ static void video_encode_example(const char *filename, int codec_id)
        fflush(stdout);
        /* prepare a dummy image */
        /* Y */
-        for(y=0;y<c->height;y++) {
-            for(x=0;x<c->width;x++) {
+        for (y = 0; y < c->height; y++) {
+            for (x = 0; x < c->width; x++) {
                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
            }
        }

        /* Cb and Cr */
-        for(y=0;y<c->height/2;y++) {
-            for(x=0;x<c->width/2;x++) {
+        for (y = 0; y < c->height/2; y++) {
+            for (x = 0; x < c->width/2; x++) {
                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
            }
@@ -467,7 +475,7 @@ static void video_encode_example(const char *filename, int codec_id)
    avcodec_close(c);
    av_free(c);
    av_freep(&frame->data[0]);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
    printf("\n");
 }

@@ -481,10 +489,10 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
    FILE *f;
    int i;

-    f=fopen(filename,"w");
-    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
-    for(i=0;i<ysize;i++)
-        fwrite(buf + i * wrap,1,xsize,f);
+    f = fopen(filename,"w");
+    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
+    for (i = 0; i < ysize; i++)
+        fwrite(buf + i * wrap, 1, xsize, f);
    fclose(f);
 }

@@ -565,14 +573,14 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }

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

    frame_count = 0;
-    for(;;) {
+    for (;;) {
        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
        if (avpkt.size == 0)
            break;
@@ -609,7 +617,7 @@ static void video_decode_example(const char *outfilename, const char *filename)

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

@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2014 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat AVIOContext API example.
+ *
+ * Make libavformat demuxer access media content through a custom
+ * AVIOContext read callback.
+ * @example avio_reading.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+#include <libavutil/file.h>
+
+struct buffer_data {
+    uint8_t *ptr;
+    size_t size; ///< size left in the buffer
+};
+
+static int read_packet(void *opaque, uint8_t *buf, int buf_size)
+{
+    struct buffer_data *bd = (struct buffer_data *)opaque;
+    buf_size = FFMIN(buf_size, bd->size);
+
+    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
+
+    /* copy internal buffer data to buf */
+    memcpy(buf, bd->ptr, buf_size);
+    bd->ptr  += buf_size;
+    bd->size -= buf_size;
+
+    return buf_size;
+}
+
+int main(int argc, char *argv[])
+{
+    AVFormatContext *fmt_ctx = NULL;
+    AVIOContext *avio_ctx = NULL;
+    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
+    size_t buffer_size, avio_ctx_buffer_size = 4096;
+    char *input_filename = NULL;
+    int ret = 0;
+    struct buffer_data bd = { 0 };
+
+    if (argc != 2) {
+        fprintf(stderr, "usage: %s input_file\n"
+                "API example program to show how to read from a custom buffer "
+                "accessed through AVIOContext.\n", argv[0]);
+        return 1;
+    }
+    input_filename = argv[1];
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+
+    /* slurp file content into buffer */
+    ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
+    if (ret < 0)
+        goto end;
+
+    /* fill opaque structure used by the AVIOContext read callback */
+    bd.ptr  = buffer;
+    bd.size = buffer_size;
+
+    if (!(fmt_ctx = avformat_alloc_context())) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
+    if (!avio_ctx_buffer) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
+                                  0, &bd, &read_packet, NULL, NULL);
+    if (!avio_ctx) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fmt_ctx->pb = avio_ctx;
+
+    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open input\n");
+        goto end;
+    }
+
+    ret = avformat_find_stream_info(fmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        goto end;
+    }
+
+    av_dump_format(fmt_ctx, 0, input_filename, 0);
+
+end:
+    avformat_close_input(&fmt_ctx);
+    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
+    if (avio_ctx) {
+        av_freep(&avio_ctx->buffer);
+        av_freep(&avio_ctx);
+    }
+    av_file_unmap(buffer, buffer_size);
+
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
@@ -22,11 +22,11 @@

 /**
 * @file
- * libavformat demuxing API use example.
+ * Demuxing and decoding example.
 *
 * Show how to use the libavformat and libavcodec API to demux and
 * decode audio and video data.
- * @example doc/examples/demuxing.c
+ * @example demuxing_decoding.c
 */

 #include <libavutil/imgutils.h>
@@ -47,25 +47,36 @@ static uint8_t *video_dst_data[4] = {NULL};
 static int      video_dst_linesize[4];
 static int video_dst_bufsize;

-static uint8_t **audio_dst_data = NULL;
-static int       audio_dst_linesize;
-static int audio_dst_bufsize;
-
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
 static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;

+/* The different ways of decoding and managing data memory. You are not
+ * supposed to support all the modes in your application but pick the one most
+ * appropriate to your needs. Look for the use of api_mode in this example to
+ * see what are the differences of API usage between them */
+enum {
+    API_MODE_OLD                  = 0, /* old method, deprecated */
+    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
+    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
+};
+
+static int api_mode = API_MODE_OLD;
+
 static int decode_packet(int *got_frame, int cached)
 {
    int ret = 0;
+    int decoded = pkt.size;
+
+    *got_frame = 0;

    if (pkt.stream_index == video_stream_idx) {
        /* decode video frame */
        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame\n");
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
            return ret;
        }

@@ -88,40 +99,40 @@ static int decode_packet(int *got_frame, int cached)
        /* decode audio frame */
        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame\n");
+            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
            return ret;
        }
+        /* Some audio decoders decode only part of the packet, and have to be
+         * called again with the remainder of the packet data.
+         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
+         * Also, some decoders might over-read the packet. */
+        decoded = FFMIN(ret, pkt.size);

        if (*got_frame) {
+            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   audio_frame_count++, frame->nb_samples,
                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));

-            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
-                                   frame->nb_samples, frame->format, 1);
-            if (ret < 0) {
-                fprintf(stderr, "Could not allocate audio buffer\n");
-                return AVERROR(ENOMEM);
-            }
-
-            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
-            audio_dst_bufsize =
-                av_samples_get_buffer_size(NULL, 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, av_frame_get_channels(frame), frame->format);
-
-            /* write to rawaudio file */
-            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
-            av_freep(&audio_dst_data[0]);
+            /* Write the raw audio data samples of the first plane. This works
+             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
+             * most audio decoders output planar audio, which uses a separate
+             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
+             * In other words, this code will write only the first audio channel
+             * in these cases.
+             * You should use libswresample or libavfilter to convert the frame
+             * to packed data. */
+            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
        }
    }

-    return ret;
+    /* If we use the new API with reference counting, we own the data and need
+     * to de-reference it when we don't use it anymore */
+    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
+        av_frame_unref(frame);
+
+    return decoded;
 }

 static int open_codec_context(int *stream_idx,
@@ -131,6 +142,7 @@ static int open_codec_context(int *stream_idx,
    AVStream *st;
    AVCodecContext *dec_ctx = NULL;
    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;

    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
    if (ret < 0) {
@@ -147,10 +159,13 @@ static int open_codec_context(int *stream_idx,
        if (!dec) {
            fprintf(stderr, "Failed to find %s codec\n",
                    av_get_media_type_string(type));
-            return ret;
+            return AVERROR(EINVAL);
        }

-        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+        /* Init the decoders, with or without reference counting */
+        if (api_mode == API_MODE_NEW_API_REF_COUNT)
+            av_dict_set(&opts, "refcounted_frames", "1", 0);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
@@ -193,15 +208,31 @@ int main (int argc, char **argv)
 {
    int ret = 0, got_frame;

-    if (argc != 4) {
-        fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
+    if (argc != 4 && argc != 5) {
+        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
+                "input_file video_output_file audio_output_file\n"
                "API example program to show how to read frames from an input file.\n"
                "This program reads frames from a file, decodes them, and writes decoded\n"
                "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n"
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
+                "If the -refcount option is specified, the program use the\n"
+                "reference counting frame system which allows keeping a copy of\n"
+                "the data for longer than one decode call. If unset, it's using\n"
+                "the classic old method.\n"
                "\n", argv[0]);
        exit(1);
    }
+    if (argc == 5) {
+        const char *mode = argv[1] + strlen("-refcount=");
+        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
+        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
+        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
+        else {
+            fprintf(stderr, "unknow mode '%s'\n", mode);
+            exit(1);
+        }
+        argv++;
+    }
    src_filename = argv[1];
    video_dst_filename = argv[2];
    audio_dst_filename = argv[3];
@@ -244,8 +275,6 @@ int main (int argc, char **argv)
    }

    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
-        int nb_planes;
-
        audio_stream = fmt_ctx->streams[audio_stream_idx];
        audio_dec_ctx = audio_stream->codec;
        audio_dst_file = fopen(audio_dst_filename, "wb");
@@ -254,15 +283,6 @@ int main (int argc, char **argv)
            ret = 1;
            goto end;
        }
-
-        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
-            audio_dec_ctx->channels : 1;
-        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
-        if (!audio_dst_data) {
-            fprintf(stderr, "Could not allocate audio data buffers\n");
-            ret = AVERROR(ENOMEM);
-            goto end;
-        }
    }

    /* dump input information to stderr */
@@ -274,7 +294,12 @@ int main (int argc, char **argv)
        goto end;
    }

-    frame = avcodec_alloc_frame();
+    /* When using the new API, you need to use the libavutil/frame.h API, while
+     * the classic frame management is available in libavcodec */
+    if (api_mode == API_MODE_OLD)
+        frame = avcodec_alloc_frame();
+    else
+        frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
@@ -293,8 +318,15 @@ int main (int argc, char **argv)

    /* read frames from the file */
    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        decode_packet(&got_frame, 0);
-        av_free_packet(&pkt);
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_free_packet(&orig_pkt);
    }

    /* flush cached frames */
@@ -314,29 +346,41 @@ int main (int argc, char **argv)
    }

    if (audio_stream) {
+        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
+        int n_channels = audio_dec_ctx->channels;
        const char *fmt;

-        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
+        if (av_sample_fmt_is_planar(sfmt)) {
+            const char *packed = av_get_sample_fmt_name(sfmt);
+            printf("Warning: the sample format the decoder produced is planar "
+                   "(%s). This example will output the first channel only.\n",
+                   packed ? packed : "?");
+            sfmt = av_get_packed_sample_fmt(sfmt);
+            n_channels = 1;
+        }
+
+        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
            goto end;
+
        printf("Play the output audio file with the command:\n"
               "ffplay -f %s -ac %d -ar %d %s\n",
-               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
+               fmt, n_channels, audio_dec_ctx->sample_rate,
               audio_dst_filename);
    }

 end:
-    if (video_dec_ctx)
-        avcodec_close(video_dec_ctx);
-    if (audio_dec_ctx)
-        avcodec_close(audio_dec_ctx);
+    avcodec_close(video_dec_ctx);
+    avcodec_close(audio_dec_ctx);
    avformat_close_input(&fmt_ctx);
    if (video_dst_file)
        fclose(video_dst_file);
    if (audio_dst_file)
        fclose(audio_dst_file);
-    av_free(frame);
+    if (api_mode == API_MODE_OLD)
+        avcodec_free_frame(&frame);
+    else
+        av_frame_free(&frame);
    av_free(video_dst_data[0]);
-    av_free(audio_dst_data);

    return ret < 0;
 }
@@ -0,0 +1,364 @@
+/*
+ * copyright (c) 2013 Andrew Kelley
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * libavfilter API usage example.
+ *
+ * @example filter_audio.c
+ * This example will generate a sine wave audio,
+ * pass it through a simple filter chain, and then compute the MD5 checksum of
+ * the output data.
+ *
+ * The filter chain it uses is:
+ * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
+ *
+ * abuffer: This provides the endpoint where you can feed the decoded samples.
+ * volume: In this example we hardcode it to 0.90.
+ * aformat: This converts the samples to the samplefreq, channel layout,
+ *          and sample format required by the audio device.
+ * abuffersink: This provides the endpoint where you can read the samples after
+ *              they have passed through the filter chain.
+ */
+
+#include <inttypes.h>
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/md5.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+#include "libavfilter/avfilter.h"
+#include "libavfilter/buffersink.h"
+#include "libavfilter/buffersrc.h"
+
+#define INPUT_SAMPLERATE     48000
+#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
+
+#define VOLUME_VAL 0.90
+
+static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
+                             AVFilterContext **sink)
+{
+    AVFilterGraph *filter_graph;
+    AVFilterContext *abuffer_ctx;
+    AVFilter        *abuffer;
+    AVFilterContext *volume_ctx;
+    AVFilter        *volume;
+    AVFilterContext *aformat_ctx;
+    AVFilter        *aformat;
+    AVFilterContext *abuffersink_ctx;
+    AVFilter        *abuffersink;
+
+    AVDictionary *options_dict = NULL;
+    uint8_t options_str[1024];
+    uint8_t ch_layout[64];
+
+    int err;
+
+    /* Create a new filtergraph, which will contain all the filters. */
+    filter_graph = avfilter_graph_alloc();
+    if (!filter_graph) {
+        fprintf(stderr, "Unable to create filter graph.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Create the abuffer filter;
+     * it will be used for feeding the data into the graph. */
+    abuffer = avfilter_get_by_name("abuffer");
+    if (!abuffer) {
+        fprintf(stderr, "Could not find the abuffer filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
+    if (!abuffer_ctx) {
+        fprintf(stderr, "Could not allocate the abuffer instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Set the filter options through the AVOptions API. */
+    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
+    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
+    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
+
+    /* Now initialize the filter; we pass NULL options, since we have already
+     * set all the options above. */
+    err = avfilter_init_str(abuffer_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffer filter.\n");
+        return err;
+    }
+
+    /* Create volume filter. */
+    volume = avfilter_get_by_name("volume");
+    if (!volume) {
+        fprintf(stderr, "Could not find the volume filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
+    if (!volume_ctx) {
+        fprintf(stderr, "Could not allocate the volume instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A different way of passing the options is as key/value pairs in a
+     * dictionary. */
+    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
+    err = avfilter_init_dict(volume_ctx, &options_dict);
+    av_dict_free(&options_dict);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the volume filter.\n");
+        return err;
+    }
+
+    /* Create the aformat filter;
+     * it ensures that the output is of the format we want. */
+    aformat = avfilter_get_by_name("aformat");
+    if (!aformat) {
+        fprintf(stderr, "Could not find the aformat filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
+    if (!aformat_ctx) {
+        fprintf(stderr, "Could not allocate the aformat instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A third way of passing the options is in a string of the form
+     * key1=value1:key2=value2.... */
+    snprintf(options_str, sizeof(options_str),
+             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
+             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
+             (uint64_t)AV_CH_LAYOUT_STEREO);
+    err = avfilter_init_str(aformat_ctx, options_str);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
+        return err;
+    }
+
+    /* Finally create the abuffersink filter;
+     * it will be used to get the filtered data out of the graph. */
+    abuffersink = avfilter_get_by_name("abuffersink");
+    if (!abuffersink) {
+        fprintf(stderr, "Could not find the abuffersink filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
+    if (!abuffersink_ctx) {
+        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* This filter takes no options. */
+    err = avfilter_init_str(abuffersink_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
+        return err;
+    }
+
+    /* Connect the filters;
+     * in this simple case the filters just form a linear chain. */
+    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
+    if (err < 0) {
+        fprintf(stderr, "Error connecting filters\n");
+        return err;
+    }
+
+    /* Configure the graph. */
+    err = avfilter_graph_config(filter_graph, NULL);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
+        return err;
+    }
+
+    *graph = filter_graph;
+    *src   = abuffer_ctx;
+    *sink  = abuffersink_ctx;
+
+    return 0;
+}
+
+/* Do something useful with the filtered data: this simple
+ * example just prints the MD5 checksum of each plane to stdout. */
+static int process_output(struct AVMD5 *md5, AVFrame *frame)
+{
+    int planar     = av_sample_fmt_is_planar(frame->format);
+    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
+    int planes     = planar ? channels : 1;
+    int bps        = av_get_bytes_per_sample(frame->format);
+    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
+    int i, j;
+
+    for (i = 0; i < planes; i++) {
+        uint8_t checksum[16];
+
+        av_md5_init(md5);
+        av_md5_sum(checksum, frame->extended_data[i], plane_size);
+
+        fprintf(stdout, "plane %d: 0x", i);
+        for (j = 0; j < sizeof(checksum); j++)
+            fprintf(stdout, "%02X", checksum[j]);
+        fprintf(stdout, "\n");
+    }
+    fprintf(stdout, "\n");
+
+    return 0;
+}
+
+/* Construct a frame of audio data to be filtered;
+ * this simple example just synthesizes a sine wave. */
+static int get_input(AVFrame *frame, int frame_num)
+{
+    int err, i, j;
+
+#define FRAME_SIZE 1024
+
+    /* Set up the frame properties and allocate the buffer for the data. */
+    frame->sample_rate    = INPUT_SAMPLERATE;
+    frame->format         = INPUT_FORMAT;
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
+    frame->nb_samples     = FRAME_SIZE;
+    frame->pts            = frame_num * FRAME_SIZE;
+
+    err = av_frame_get_buffer(frame, 0);
+    if (err < 0)
+        return err;
+
+    /* Fill the data for each channel. */
+    for (i = 0; i < 5; i++) {
+        float *data = (float*)frame->extended_data[i];
+
+        for (j = 0; j < frame->nb_samples; j++)
+            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
+    }
+
+    return 0;
+}
+
+int main(int argc, char *argv[])
+{
+    struct AVMD5 *md5;
+    AVFilterGraph *graph;
+    AVFilterContext *src, *sink;
+    AVFrame *frame;
+    uint8_t errstr[1024];
+    float duration;
+    int err, nb_frames, i;
+
+    if (argc < 2) {
+        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
+        return 1;
+    }
+
+    duration  = atof(argv[1]);
+    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
+    if (nb_frames <= 0) {
+        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
+        return 1;
+    }
+
+    avfilter_register_all();
+
+    /* Allocate the frame we will be using to store the data. */
+    frame  = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Error allocating the frame\n");
+        return 1;
+    }
+
+    md5 = av_md5_alloc();
+    if (!md5) {
+        fprintf(stderr, "Error allocating the MD5 context\n");
+        return 1;
+    }
+
+    /* Set up the filtergraph. */
+    err = init_filter_graph(&graph, &src, &sink);
+    if (err < 0) {
+        fprintf(stderr, "Unable to init filter graph:");
+        goto fail;
+    }
+
+    /* the main filtering loop */
+    for (i = 0; i < nb_frames; i++) {
+        /* get an input frame to be filtered */
+        err = get_input(frame, i);
+        if (err < 0) {
+            fprintf(stderr, "Error generating input frame:");
+            goto fail;
+        }
+
+        /* Send the frame to the input of the filtergraph. */
+        err = av_buffersrc_add_frame(src, frame);
+        if (err < 0) {
+            av_frame_unref(frame);
+            fprintf(stderr, "Error submitting the frame to the filtergraph:");
+            goto fail;
+        }
+
+        /* Get all the filtered output that is available. */
+        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
+            /* now do something with our filtered frame */
+            err = process_output(md5, frame);
+            if (err < 0) {
+                fprintf(stderr, "Error processing the filtered frame:");
+                goto fail;
+            }
+            av_frame_unref(frame);
+        }
+
+        if (err == AVERROR(EAGAIN)) {
+            /* Need to feed more frames in. */
+            continue;
+        } else if (err == AVERROR_EOF) {
+            /* Nothing more to do, finish. */
+            break;
+        } else if (err < 0) {
+            /* An error occurred. */
+            fprintf(stderr, "Error filtering the data:");
+            goto fail;
+        }
+    }
+
+    avfilter_graph_free(&graph);
+    av_frame_free(&frame);
+    av_freep(&md5);
+
+    return 0;
+
+fail:
+    av_strerror(err, errstr, sizeof(errstr));
+    fprintf(stderr, "%s\n", errstr);
+    return 1;
+}
@@ -25,7 +25,7 @@
 /**
 * @file
 * API example for audio decoding and filtering
- * @example doc/examples/filtering_audio.c
+ * @example filtering_audio.c
 */

 #include <unistd.h>
@@ -36,9 +36,10 @@
 #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 *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
+static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -70,6 +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) {
@@ -83,17 +85,22 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
    char args[512];
-    int ret;
+    int ret = 0;
    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;
+    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    static const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }

    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
    if (!dec_ctx->channel_layout)
@@ -106,18 +113,36 @@ static int init_filters(const char *filters_descr)
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
+        goto end;
    }

    /* 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;
+        goto end;
+    }
+
+    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");
+        goto end;
+    }
+
+    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");
+        goto end;
+    }
+
+    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");
+        goto end;
    }

    /* Endpoints for the filter graph. */
@@ -131,12 +156,12 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

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

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
+        goto end;

    /* Print summary of the sink buffer
     * Note: args buffer is reused to store channel layout string */
@@ -147,14 +172,17 @@ static int init_filters(const char *filters_descr)
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
           args);

-    return 0;
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }

-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) {
@@ -168,11 +196,12 @@ static void print_samplesref(AVFilterBufferRef *samplesref)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket packet;
-    AVFrame *frame = avcodec_alloc_frame();
+    AVPacket packet0, packet;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;

-    if (!frame) {
+    if (!frame || !filt_frame) {
        perror("Could not allocate frame");
        exit(1);
    }
@@ -181,7 +210,6 @@ int main(int argc, char **argv)
        exit(1);
    }

-    avcodec_register_all();
    av_register_all();
    avfilter_register_all();

@@ -191,54 +219,60 @@ int main(int argc, char **argv)
        goto end;

    /* read all packets */
+    packet0.data = NULL;
+    packet.data = NULL;
    while (1) {
-        AVFilterBufferRef *samplesref;
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
+        if (!packet0.data) {
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+                break;
+            packet0 = packet;
+        }

        if (packet.stream_index == audio_stream_index) {
-            avcodec_get_frame_defaults(frame);
            got_frame = 0;
            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                continue;
            }
+            packet.size -= ret;
+            packet.data += ret;

            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);
-                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
-                    if(ret < 0)
+                    if (ret < 0)
                        goto end;
-                    if (samplesref) {
-                        print_samplesref(samplesref);
-                        avfilter_unref_bufferp(&samplesref);
-                    }
+                    print_frame(filt_frame);
+                    av_frame_unref(filt_frame);
                }
            }
+
+            if (packet.size <= 0)
+                av_free_packet(&packet0);
+        } else {
+            /* discard non-wanted packets */
+            av_free_packet(&packet0);
        }
-        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
-    av_freep(&frame);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }

@@ -24,7 +24,7 @@
 /**
 * @file
 * API example for decoding and filtering
- * @example doc/examples/filtering_video.c
+ * @example filtering_video.c
 */

 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -36,6 +36,7 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>

 const char *filter_descr = "scale=78:24";

@@ -70,6 +71,7 @@ static int open_input_file(const char *filename)
    }
    video_stream_index = ret;
    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);

    /* init the video decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -83,15 +85,18 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
    char args[512];
-    int ret;
+    int ret = 0;
    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
-    AVBufferSinkParams *buffersink_params;

    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }

    /* buffer video source: the decoded frames from the decoder will be inserted here. */
    snprintf(args, sizeof(args),
@@ -104,18 +109,22 @@ static int init_filters(const char *filters_descr)
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        return ret;
+        goto end;
    }

    /* buffer video sink: to terminate the filter chain. */
-    buffersink_params = av_buffersink_params_alloc();
-    buffersink_params->pixel_fmts = pix_fmts;
    ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, buffersink_params, filter_graph);
-    av_free(buffersink_params);
+                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        return ret;
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
+                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
+        goto end;
    }

    /* Endpoints for the filter graph. */
@@ -129,42 +138,47 @@ 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;
+        goto end;

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-    return 0;
+        goto end;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }

-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);
 }
@@ -173,10 +187,11 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame *frame = avcodec_alloc_frame();
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;

-    if (!frame) {
+    if (!frame || !filt_frame) {
        perror("Could not allocate frame");
        exit(1);
    }
@@ -185,7 +200,6 @@ int main(int argc, char **argv)
        exit(1);
    }

-    avcodec_register_all();
    av_register_all();
    avfilter_register_all();

@@ -196,12 +210,10 @@ 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);
            got_frame = 0;
            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
@@ -213,39 +225,35 @@ int main(int argc, char **argv)
                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);
                }
+                av_frame_unref(frame);
            }
        }
        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
-    av_freep(&frame);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }

@@ -23,7 +23,7 @@
 /**
 * @file
 * Shows how the metadata API can be used in application programs.
- * @example doc/examples/metadata.c
+ * @example metadata.c
 */

 #include <stdio.h>
@@ -24,9 +24,9 @@
 * @file
 * libavformat API example.
 *
- * Output a media file in any supported libavformat format.
- * The default codecs are used.
- * @example doc/examples/muxing.c
+ * Output a media file in any supported libavformat format. The default
+ * codecs are used.
+ * @example muxing.c
 */

 #include <stdlib.h>
@@ -34,24 +34,44 @@
 #include <string.h>
 #include <math.h>

+#include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
+#include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
+#include <libswresample/swresample.h>

-/* 5 seconds stream duration */
-#define STREAM_DURATION   200.0
+static int audio_is_eof, video_is_eof;
+
+#define STREAM_DURATION   10.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
-#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */

 static int sws_flags = SWS_BICUBIC;

-/**************************************************************/
-/* audio output */
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;

-static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
+    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
+{
+    /* rescale output packet timestamp values from codec to stream timebase */
+    pkt->pts = av_rescale_q_rnd(pkt->pts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->dts = av_rescale_q_rnd(pkt->dts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->duration = av_rescale_q(pkt->duration, *time_base, st->time_base);
+    pkt->stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    log_packet(fmt_ctx, pkt);
+    return av_interleaved_write_frame(fmt_ctx, pkt);
+}

 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
@@ -78,8 +98,8 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,

    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
-        st->id = 1;
-        c->sample_fmt  = AV_SAMPLE_FMT_S16;
+        c->sample_fmt  = (*codec)->sample_fmts ?
+            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
        c->channels    = 2;
@@ -127,8 +147,19 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 /* audio output */

 static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
+
+AVFrame *audio_frame;
+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;
+int samples_count;
+
+struct SwrContext *swr_ctx = NULL;

 static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
@@ -137,6 +168,13 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)

    c = st->codec;

+    /* allocate and init a re-usable frame */
+    audio_frame = av_frame_alloc();
+    if (!audio_frame) {
+        fprintf(stderr, "Could not allocate audio frame\n");
+        exit(1);
+    }
+
    /* open it */
    ret = avcodec_open2(c, codec, NULL);
    if (ret < 0) {
@@ -150,17 +188,54 @@ 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);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate audio samples buffer\n");
+    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, AV_SAMPLE_FMT_S16, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\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;
+
+    /* 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);
+        }
+
+        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);
+        }
+    } else {
+        dst_samples_data = src_samples_data;
+    }
+    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
@@ -180,50 +255,83 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
    }
 }

-static void write_audio_frame(AVFormatContext *oc, AVStream *st)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet, ret;
+    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);
+    if (!flush) {
+        get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);

-    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+        /* 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_nb_samples = src_nb_samples;
+        }
+
+        audio_frame->nb_samples = dst_nb_samples;
+        audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
+                                 dst_samples_data[0], dst_samples_size, 0);
+        samples_count += dst_nb_samples;
+    }
+
+    ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
    if (ret < 0) {
        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
        exit(1);
    }

-    if (!got_packet)
+    if (!got_packet) {
+        if (flush)
+            audio_is_eof = 1;
        return;
+    }

-    pkt.stream_index = st->index;
-
-    /* Write the compressed frame to the media file. */
-    ret = av_interleaved_write_frame(oc, &pkt);
-    if (ret != 0) {
+    ret = write_frame(oc, &c->time_base, st, &pkt);
+    if (ret < 0) {
        fprintf(stderr, "Error while writing audio frame: %s\n",
                av_err2str(ret));
        exit(1);
    }
-    avcodec_free_frame(&frame);
 }

 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
    avcodec_close(st->codec);
-
-    av_free(samples);
+    if (dst_samples_data != src_samples_data) {
+        av_free(dst_samples_data[0]);
+        av_free(dst_samples_data);
+    }
+    av_free(src_samples_data[0]);
+    av_free(src_samples_data);
+    av_frame_free(&audio_frame);
 }

 /**************************************************************/
@@ -246,11 +354,14 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    }

    /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
+    frame->format = c->pix_fmt;
+    frame->width = c->width;
+    frame->height = c->height;

    /* Allocate the encoded raw picture. */
    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
@@ -297,17 +408,13 @@ static void fill_yuv_image(AVPicture *pict, int frame_index,
    }
 }

-static void write_video_frame(AVFormatContext *oc, AVStream *st)
+static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
    int ret;
    static struct SwsContext *sws_ctx;
    AVCodecContext *c = st->codec;

-    if (frame_count >= STREAM_NB_FRAMES) {
-        /* No more frames to compress. The codec has a latency of a few
-         * frames if using B-frames, so we get the last frames by
-         * passing the same picture again. */
-    } else {
+    if (!flush) {
        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 */
@@ -330,7 +437,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
        }
    }

-    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+    if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) {
        /* Raw video case - directly store the picture in the packet */
        AVPacket pkt;
        av_init_packet(&pkt);
@@ -342,34 +449,29 @@ 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 */
+        frame->pts = frame_count;
+        ret = avcodec_encode_video2(c, &pkt, flush ? NULL : frame, &got_packet);
        if (ret < 0) {
            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
            exit(1);
        }
-
        /* If size is zero, it means the image was buffered. */
-        if (got_output) {
-            if (c->coded_frame->key_frame)
-                pkt.flags |= AV_PKT_FLAG_KEY;

-            pkt.stream_index = st->index;
-
-            /* Write the compressed frame to the media file. */
-            ret = av_interleaved_write_frame(oc, &pkt);
+        if (got_packet) {
+            ret = write_frame(oc, &c->time_base, st, &pkt);
        } else {
+            if (flush)
+                video_is_eof = 1;
            ret = 0;
        }
    }
-    if (ret != 0) {
+
+    if (ret < 0) {
        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
        exit(1);
    }
@@ -381,7 +483,7 @@ static void close_video(AVFormatContext *oc, AVStream *st)
    avcodec_close(st->codec);
    av_free(src_picture.data[0]);
    av_free(dst_picture.data[0]);
-    av_free(frame);
+    av_frame_free(&frame);
 }

 /**************************************************************/
@@ -394,8 +496,8 @@ int main(int argc, char **argv)
    AVFormatContext *oc;
    AVStream *audio_st, *video_st;
    AVCodec *audio_codec, *video_codec;
-    double audio_pts, video_pts;
-    int ret;
+    double audio_time, video_time;
+    int flush, ret;

    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();
@@ -419,9 +521,9 @@ int main(int argc, char **argv)
        printf("Could not deduce output format from file extension: using MPEG.\n");
        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
    }
-    if (!oc) {
+    if (!oc)
        return 1;
-    }
+
    fmt = oc->oformat;

    /* Add the audio and video streams using the default format codecs
@@ -429,12 +531,10 @@ int main(int argc, char **argv)
    video_st = NULL;
    audio_st = NULL;

-    if (fmt->video_codec != AV_CODEC_ID_NONE) {
+    if (fmt->video_codec != AV_CODEC_ID_NONE)
        video_st = add_stream(oc, &video_codec, fmt->video_codec);
-    }
-    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
+    if (fmt->audio_codec != AV_CODEC_ID_NONE)
        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
-    }

    /* Now that all the parameters are set, we can open the audio and
     * video codecs and allocate the necessary encode buffers. */
@@ -463,31 +563,23 @@ int main(int argc, char **argv)
        return 1;
    }

-    if (frame)
-        frame->pts = 0;
-    for (;;) {
+    flush = 0;
+    while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
        /* 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_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY;
+        video_time = (video_st && !video_is_eof) ? video_st->pts.val * av_q2d(video_st->time_base) : INFINITY;

-        if (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
-        else
-            video_pts = 0.0;
-
-        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
-            (!video_st || video_pts >= STREAM_DURATION))
-            break;
+        if (!flush &&
+            (!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION)) {
+            flush = 1;
+        }

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

@@ -0,0 +1,164 @@
+/*
+ * Copyright (c) 2013 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat/libavcodec demuxing and muxing API example.
+ *
+ * Remux streams from one container format to another.
+ * @example remuxing.c
+ */
+
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           tag,
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+int main(int argc, char **argv)
+{
+    AVOutputFormat *ofmt = NULL;
+    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
+    AVPacket pkt;
+    const char *in_filename, *out_filename;
+    int ret, i;
+
+    if (argc < 3) {
+        printf("usage: %s input output\n"
+               "API example program to remux a media file with libavformat and libavcodec.\n"
+               "The output format is guessed according to the file extension.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_filename  = argv[1];
+    out_filename = argv[2];
+
+    av_register_all();
+
+    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
+        fprintf(stderr, "Could not open input file '%s'", in_filename);
+        goto end;
+    }
+
+    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
+        fprintf(stderr, "Failed to retrieve input stream information");
+        goto end;
+    }
+
+    av_dump_format(ifmt_ctx, 0, in_filename, 0);
+
+    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
+    if (!ofmt_ctx) {
+        fprintf(stderr, "Could not create output context\n");
+        ret = AVERROR_UNKNOWN;
+        goto end;
+    }
+
+    ofmt = ofmt_ctx->oformat;
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *in_stream = ifmt_ctx->streams[i];
+        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
+        if (!out_stream) {
+            fprintf(stderr, "Failed allocating output stream\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
+        if (ret < 0) {
+            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
+            goto end;
+        }
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
+    }
+    av_dump_format(ofmt_ctx, 0, out_filename, 1);
+
+    if (!(ofmt->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            fprintf(stderr, "Could not open output file '%s'", out_filename);
+            goto end;
+        }
+    }
+
+    ret = avformat_write_header(ofmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred when opening output file\n");
+        goto end;
+    }
+
+    while (1) {
+        AVStream *in_stream, *out_stream;
+
+        ret = av_read_frame(ifmt_ctx, &pkt);
+        if (ret < 0)
+            break;
+
+        in_stream  = ifmt_ctx->streams[pkt.stream_index];
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
+
+        log_packet(ifmt_ctx, &pkt, "in");
+
+        /* copy packet */
+        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
+        pkt.pos = -1;
+        log_packet(ofmt_ctx, &pkt, "out");
+
+        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error muxing packet\n");
+            break;
+        }
+        av_free_packet(&pkt);
+    }
+
+    av_write_trailer(ofmt_ctx);
+end:
+
+    avformat_close_input(&ifmt_ctx);
+
+    /* close output */
+    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
+        avio_close(ofmt_ctx->pb);
+    avformat_free_context(ofmt_ctx);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
@@ -21,7 +21,7 @@
 */

 /**
- * @example doc/examples/resampling_audio.c
+ * @example resampling_audio.c
 * libswresample API use example.
 */

@@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 /**
 * 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)
+static 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;
@@ -78,18 +78,6 @@ void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate,
    }
 }

-int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
-                                    int nb_samples, enum AVSampleFormat sample_fmt, int align)
-{
-    int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
-
-    *data = av_malloc(sizeof(*data) * nb_planes);
-    if (!*data)
-        return AVERROR(ENOMEM);
-    return av_samples_alloc(*data, linesize, nb_channels,
-                            nb_samples, sample_fmt, align);
-}
-
 int main(int argc, char **argv)
 {
    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
@@ -149,8 +137,8 @@ int main(int argc, char **argv)
    /* allocate source and destination samples buffers */

    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
-    ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
-                                       src_nb_samples, src_sample_fmt, 0);
+    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;
@@ -164,8 +152,8 @@ int main(int argc, char **argv)

    /* buffer is going to be directly written to a rawaudio file, no alignment */
    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
-    ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
-                                       dst_nb_samples, dst_sample_fmt, 0);
+    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;
@@ -196,6 +184,10 @@ int main(int argc, char **argv)
        }
        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                 ret, dst_sample_fmt, 1);
+        if (dst_bufsize < 0) {
+            fprintf(stderr, "Could not get sample buffer size\n");
+            goto end;
+        }
        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);
@@ -23,7 +23,7 @@
 /**
 * @file
 * libswscale API use example.
- * @example doc/examples/scaling_video.c
+ * @example scaling_video.c
 */

 #include <libavutil/imgutils.h>
@@ -0,0 +1,755 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * simple audio converter
+ *
+ * @example transcode_aac.c
+ * Convert an input audio file to AAC in an MP4 container using FFmpeg.
+ * @author Andreas Unterweger (dustsigns@gmail.com)
+ */
+
+#include <stdio.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/frame.h"
+#include "libavutil/opt.h"
+
+#include "libswresample/swresample.h"
+
+/** The output bit rate in kbit/s */
+#define OUTPUT_BIT_RATE 48000
+/** The number of output channels */
+#define OUTPUT_CHANNELS 2
+/** The audio sample output format */
+#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
+
+/**
+ * Convert an error code into a text message.
+ * @param error Error code to be converted
+ * @return Corresponding error text (not thread-safe)
+ */
+static char *const get_error_text(const int error)
+{
+    static char error_buffer[255];
+    av_strerror(error, error_buffer, sizeof(error_buffer));
+    return error_buffer;
+}
+
+/** Open an input file and the required decoder. */
+static int open_input_file(const char *filename,
+                           AVFormatContext **input_format_context,
+                           AVCodecContext **input_codec_context)
+{
+    AVCodec *input_codec;
+    int error;
+
+    /** Open the input file to read from it. */
+    if ((error = avformat_open_input(input_format_context, filename, NULL,
+                                     NULL)) < 0) {
+        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        *input_format_context = NULL;
+        return error;
+    }
+
+    /** Get information on the input file (number of streams etc.). */
+    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
+        fprintf(stderr, "Could not open find stream info (error '%s')\n",
+                get_error_text(error));
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Make sure that there is only one stream in the input file. */
+    if ((*input_format_context)->nb_streams != 1) {
+        fprintf(stderr, "Expected one audio input stream, but found %d\n",
+                (*input_format_context)->nb_streams);
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** Find a decoder for the audio stream. */
+    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
+        fprintf(stderr, "Could not find input codec\n");
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** Open the decoder for the audio stream to use it later. */
+    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
+                               input_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open input codec (error '%s')\n",
+                get_error_text(error));
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Save the decoder context for easier access later. */
+    *input_codec_context = (*input_format_context)->streams[0]->codec;
+
+    return 0;
+}
+
+/**
+ * Open an output file and the required encoder.
+ * Also set some basic encoder parameters.
+ * Some of these parameters are based on the input file's parameters.
+ */
+static int open_output_file(const char *filename,
+                            AVCodecContext *input_codec_context,
+                            AVFormatContext **output_format_context,
+                            AVCodecContext **output_codec_context)
+{
+    AVIOContext *output_io_context = NULL;
+    AVStream *stream               = NULL;
+    AVCodec *output_codec          = NULL;
+    int error;
+
+    /** Open the output file to write to it. */
+    if ((error = avio_open(&output_io_context, filename,
+                           AVIO_FLAG_WRITE)) < 0) {
+        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        return error;
+    }
+
+    /** Create a new format context for the output container format. */
+    if (!(*output_format_context = avformat_alloc_context())) {
+        fprintf(stderr, "Could not allocate output format context\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /** Associate the output file (pointer) with the container format context. */
+    (*output_format_context)->pb = output_io_context;
+
+    /** Guess the desired container format based on the file extension. */
+    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
+                                                              NULL))) {
+        fprintf(stderr, "Could not find output file format\n");
+        goto cleanup;
+    }
+
+    av_strlcpy((*output_format_context)->filename, filename,
+               sizeof((*output_format_context)->filename));
+
+    /** Find the encoder to be used by its name. */
+    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
+        fprintf(stderr, "Could not find an AAC encoder.\n");
+        goto cleanup;
+    }
+
+    /** Create a new audio stream in the output file container. */
+    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
+        fprintf(stderr, "Could not create new stream\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    /** Save the encoder context for easiert access later. */
+    *output_codec_context = stream->codec;
+
+    /**
+     * Set the basic encoder parameters.
+     * The input file's sample rate is used to avoid a sample rate conversion.
+     */
+    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
+    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
+    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
+    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
+    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
+
+    /**
+     * Some container formats (like MP4) require global headers to be present
+     * Mark the encoder so that it behaves accordingly.
+     */
+    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
+        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    /** Open the encoder for the audio stream to use it later. */
+    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open output codec (error '%s')\n",
+                get_error_text(error));
+        goto cleanup;
+    }
+
+    return 0;
+
+cleanup:
+    avio_close((*output_format_context)->pb);
+    avformat_free_context(*output_format_context);
+    *output_format_context = NULL;
+    return error < 0 ? error : AVERROR_EXIT;
+}
+
+/** Initialize one data packet for reading or writing. */
+static void init_packet(AVPacket *packet)
+{
+    av_init_packet(packet);
+    /** Set the packet data and size so that it is recognized as being empty. */
+    packet->data = NULL;
+    packet->size = 0;
+}
+
+/** Initialize one audio frame for reading from the input file */
+static int init_input_frame(AVFrame **frame)
+{
+    if (!(*frame = av_frame_alloc())) {
+        fprintf(stderr, "Could not allocate input frame\n");
+        return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+/**
+ * Initialize the audio resampler based on the input and output codec settings.
+ * If the input and output sample formats differ, a conversion is required
+ * libswresample takes care of this, but requires initialization.
+ */
+static int init_resampler(AVCodecContext *input_codec_context,
+                          AVCodecContext *output_codec_context,
+                          SwrContext **resample_context)
+{
+        int error;
+
+        /**
+         * Create a resampler context for the conversion.
+         * Set the conversion parameters.
+         * Default channel layouts based on the number of channels
+         * are assumed for simplicity (they are sometimes not detected
+         * properly by the demuxer and/or decoder).
+         */
+        *resample_context = swr_alloc_set_opts(NULL,
+                                              av_get_default_channel_layout(output_codec_context->channels),
+                                              output_codec_context->sample_fmt,
+                                              output_codec_context->sample_rate,
+                                              av_get_default_channel_layout(input_codec_context->channels),
+                                              input_codec_context->sample_fmt,
+                                              input_codec_context->sample_rate,
+                                              0, NULL);
+        if (!*resample_context) {
+            fprintf(stderr, "Could not allocate resample context\n");
+            return AVERROR(ENOMEM);
+        }
+        /**
+        * Perform a sanity check so that the number of converted samples is
+        * not greater than the number of samples to be converted.
+        * If the sample rates differ, this case has to be handled differently
+        */
+        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
+
+        /** Open the resampler with the specified parameters. */
+        if ((error = swr_init(*resample_context)) < 0) {
+            fprintf(stderr, "Could not open resample context\n");
+            swr_free(resample_context);
+            return error;
+        }
+    return 0;
+}
+
+/** Initialize a FIFO buffer for the audio samples to be encoded. */
+static int init_fifo(AVAudioFifo **fifo)
+{
+    /** Create the FIFO buffer based on the specified output sample format. */
+    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
+        fprintf(stderr, "Could not allocate FIFO\n");
+        return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+/** Write the header of the output file container. */
+static int write_output_file_header(AVFormatContext *output_format_context)
+{
+    int error;
+    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
+        fprintf(stderr, "Could not write output file header (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+    return 0;
+}
+
+/** Decode one audio frame from the input file. */
+static int decode_audio_frame(AVFrame *frame,
+                              AVFormatContext *input_format_context,
+                              AVCodecContext *input_codec_context,
+                              int *data_present, int *finished)
+{
+    /** Packet used for temporary storage. */
+    AVPacket input_packet;
+    int error;
+    init_packet(&input_packet);
+
+    /** Read one audio frame from the input file into a temporary packet. */
+    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
+        /** If we are the the end of the file, flush the decoder below. */
+        if (error == AVERROR_EOF)
+            *finished = 1;
+        else {
+            fprintf(stderr, "Could not read frame (error '%s')\n",
+                    get_error_text(error));
+            return error;
+        }
+    }
+
+    /**
+     * Decode the audio frame stored in the temporary packet.
+     * The input audio stream decoder is used to do this.
+     * If we are at the end of the file, pass an empty packet to the decoder
+     * to flush it.
+     */
+    if ((error = avcodec_decode_audio4(input_codec_context, frame,
+                                       data_present, &input_packet)) < 0) {
+        fprintf(stderr, "Could not decode frame (error '%s')\n",
+                get_error_text(error));
+        av_free_packet(&input_packet);
+        return error;
+    }
+
+    /**
+     * If the decoder has not been flushed completely, we are not finished,
+     * so that this function has to be called again.
+     */
+    if (*finished && *data_present)
+        *finished = 0;
+    av_free_packet(&input_packet);
+    return 0;
+}
+
+/**
+ * Initialize a temporary storage for the specified number of audio samples.
+ * The conversion requires temporary storage due to the different format.
+ * The number of audio samples to be allocated is specified in frame_size.
+ */
+static int init_converted_samples(uint8_t ***converted_input_samples,
+                                  AVCodecContext *output_codec_context,
+                                  int frame_size)
+{
+    int error;
+
+    /**
+     * Allocate as many pointers as there are audio channels.
+     * Each pointer will later point to the audio samples of the corresponding
+     * channels (although it may be NULL for interleaved formats).
+     */
+    if (!(*converted_input_samples = calloc(output_codec_context->channels,
+                                            sizeof(**converted_input_samples)))) {
+        fprintf(stderr, "Could not allocate converted input sample pointers\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /**
+     * Allocate memory for the samples of all channels in one consecutive
+     * block for convenience.
+     */
+    if ((error = av_samples_alloc(*converted_input_samples, NULL,
+                                  output_codec_context->channels,
+                                  frame_size,
+                                  output_codec_context->sample_fmt, 0)) < 0) {
+        fprintf(stderr,
+                "Could not allocate converted input samples (error '%s')\n",
+                get_error_text(error));
+        av_freep(&(*converted_input_samples)[0]);
+        free(*converted_input_samples);
+        return error;
+    }
+    return 0;
+}
+
+/**
+ * Convert the input audio samples into the output sample format.
+ * The conversion happens on a per-frame basis, the size of which is specified
+ * by frame_size.
+ */
+static int convert_samples(const uint8_t **input_data,
+                           uint8_t **converted_data, const int frame_size,
+                           SwrContext *resample_context)
+{
+    int error;
+
+    /** Convert the samples using the resampler. */
+    if ((error = swr_convert(resample_context,
+                             converted_data, frame_size,
+                             input_data    , frame_size)) < 0) {
+        fprintf(stderr, "Could not convert input samples (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+
+    return 0;
+}
+
+/** Add converted input audio samples to the FIFO buffer for later processing. */
+static int add_samples_to_fifo(AVAudioFifo *fifo,
+                               uint8_t **converted_input_samples,
+                               const int frame_size)
+{
+    int error;
+
+    /**
+     * Make the FIFO as large as it needs to be to hold both,
+     * the old and the new samples.
+     */
+    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
+        fprintf(stderr, "Could not reallocate FIFO\n");
+        return error;
+    }
+
+    /** Store the new samples in the FIFO buffer. */
+    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
+                            frame_size) < frame_size) {
+        fprintf(stderr, "Could not write data to FIFO\n");
+        return AVERROR_EXIT;
+    }
+    return 0;
+}
+
+/**
+ * Read one audio frame from the input file, decodes, converts and stores
+ * it in the FIFO buffer.
+ */
+static int read_decode_convert_and_store(AVAudioFifo *fifo,
+                                         AVFormatContext *input_format_context,
+                                         AVCodecContext *input_codec_context,
+                                         AVCodecContext *output_codec_context,
+                                         SwrContext *resampler_context,
+                                         int *finished)
+{
+    /** Temporary storage of the input samples of the frame read from the file. */
+    AVFrame *input_frame = NULL;
+    /** Temporary storage for the converted input samples. */
+    uint8_t **converted_input_samples = NULL;
+    int data_present;
+    int ret = AVERROR_EXIT;
+
+    /** Initialize temporary storage for one input frame. */
+    if (init_input_frame(&input_frame))
+        goto cleanup;
+    /** Decode one frame worth of audio samples. */
+    if (decode_audio_frame(input_frame, input_format_context,
+                           input_codec_context, &data_present, finished))
+        goto cleanup;
+    /**
+     * If we are at the end of the file and there are no more samples
+     * in the decoder which are delayed, we are actually finished.
+     * This must not be treated as an error.
+     */
+    if (*finished && !data_present) {
+        ret = 0;
+        goto cleanup;
+    }
+    /** If there is decoded data, convert and store it */
+    if (data_present) {
+        /** Initialize the temporary storage for the converted input samples. */
+        if (init_converted_samples(&converted_input_samples, output_codec_context,
+                                   input_frame->nb_samples))
+            goto cleanup;
+
+        /**
+         * Convert the input samples to the desired output sample format.
+         * This requires a temporary storage provided by converted_input_samples.
+         */
+        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
+                            input_frame->nb_samples, resampler_context))
+            goto cleanup;
+
+        /** Add the converted input samples to the FIFO buffer for later processing. */
+        if (add_samples_to_fifo(fifo, converted_input_samples,
+                                input_frame->nb_samples))
+            goto cleanup;
+        ret = 0;
+    }
+    ret = 0;
+
+cleanup:
+    if (converted_input_samples) {
+        av_freep(&converted_input_samples[0]);
+        free(converted_input_samples);
+    }
+    av_frame_free(&input_frame);
+
+    return ret;
+}
+
+/**
+ * Initialize one input frame for writing to the output file.
+ * The frame will be exactly frame_size samples large.
+ */
+static int init_output_frame(AVFrame **frame,
+                             AVCodecContext *output_codec_context,
+                             int frame_size)
+{
+    int error;
+
+    /** Create a new frame to store the audio samples. */
+    if (!(*frame = av_frame_alloc())) {
+        fprintf(stderr, "Could not allocate output frame\n");
+        return AVERROR_EXIT;
+    }
+
+    /**
+     * Set the frame's parameters, especially its size and format.
+     * av_frame_get_buffer needs this to allocate memory for the
+     * audio samples of the frame.
+     * Default channel layouts based on the number of channels
+     * are assumed for simplicity.
+     */
+    (*frame)->nb_samples     = frame_size;
+    (*frame)->channel_layout = output_codec_context->channel_layout;
+    (*frame)->format         = output_codec_context->sample_fmt;
+    (*frame)->sample_rate    = output_codec_context->sample_rate;
+
+    /**
+     * Allocate the samples of the created frame. This call will make
+     * sure that the audio frame can hold as many samples as specified.
+     */
+    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
+        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
+                get_error_text(error));
+        av_frame_free(frame);
+        return error;
+    }
+
+    return 0;
+}
+
+/** Encode one frame worth of audio to the output file. */
+static int encode_audio_frame(AVFrame *frame,
+                              AVFormatContext *output_format_context,
+                              AVCodecContext *output_codec_context,
+                              int *data_present)
+{
+    /** Packet used for temporary storage. */
+    AVPacket output_packet;
+    int error;
+    init_packet(&output_packet);
+
+    /**
+     * Encode the audio frame and store it in the temporary packet.
+     * The output audio stream encoder is used to do this.
+     */
+    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
+                                       frame, data_present)) < 0) {
+        fprintf(stderr, "Could not encode frame (error '%s')\n",
+                get_error_text(error));
+        av_free_packet(&output_packet);
+        return error;
+    }
+
+    /** Write one audio frame from the temporary packet to the output file. */
+    if (*data_present) {
+        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
+            fprintf(stderr, "Could not write frame (error '%s')\n",
+                    get_error_text(error));
+            av_free_packet(&output_packet);
+            return error;
+        }
+
+        av_free_packet(&output_packet);
+    }
+
+    return 0;
+}
+
+/**
+ * Load one audio frame from the FIFO buffer, encode and write it to the
+ * output file.
+ */
+static int load_encode_and_write(AVAudioFifo *fifo,
+                                 AVFormatContext *output_format_context,
+                                 AVCodecContext *output_codec_context)
+{
+    /** Temporary storage of the output samples of the frame written to the file. */
+    AVFrame *output_frame;
+    /**
+     * Use the maximum number of possible samples per frame.
+     * If there is less than the maximum possible frame size in the FIFO
+     * buffer use this number. Otherwise, use the maximum possible frame size
+     */
+    const int frame_size = FFMIN(av_audio_fifo_size(fifo),
+                                 output_codec_context->frame_size);
+    int data_written;
+
+    /** Initialize temporary storage for one output frame. */
+    if (init_output_frame(&output_frame, output_codec_context, frame_size))
+        return AVERROR_EXIT;
+
+    /**
+     * Read as many samples from the FIFO buffer as required to fill the frame.
+     * The samples are stored in the frame temporarily.
+     */
+    if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
+        fprintf(stderr, "Could not read data from FIFO\n");
+        av_frame_free(&output_frame);
+        return AVERROR_EXIT;
+    }
+
+    /** Encode one frame worth of audio samples. */
+    if (encode_audio_frame(output_frame, output_format_context,
+                           output_codec_context, &data_written)) {
+        av_frame_free(&output_frame);
+        return AVERROR_EXIT;
+    }
+    av_frame_free(&output_frame);
+    return 0;
+}
+
+/** Write the trailer of the output file container. */
+static int write_output_file_trailer(AVFormatContext *output_format_context)
+{
+    int error;
+    if ((error = av_write_trailer(output_format_context)) < 0) {
+        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+    return 0;
+}
+
+/** Convert an audio file to an AAC file in an MP4 container. */
+int main(int argc, char **argv)
+{
+    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
+    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
+    SwrContext *resample_context = NULL;
+    AVAudioFifo *fifo = NULL;
+    int ret = AVERROR_EXIT;
+
+    if (argc < 3) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        exit(1);
+    }
+
+    /** Register all codecs and formats so that they can be used. */
+    av_register_all();
+    /** Open the input file for reading. */
+    if (open_input_file(argv[1], &input_format_context,
+                        &input_codec_context))
+        goto cleanup;
+    /** Open the output file for writing. */
+    if (open_output_file(argv[2], input_codec_context,
+                         &output_format_context, &output_codec_context))
+        goto cleanup;
+    /** Initialize the resampler to be able to convert audio sample formats. */
+    if (init_resampler(input_codec_context, output_codec_context,
+                       &resample_context))
+        goto cleanup;
+    /** Initialize the FIFO buffer to store audio samples to be encoded. */
+    if (init_fifo(&fifo))
+        goto cleanup;
+    /** Write the header of the output file container. */
+    if (write_output_file_header(output_format_context))
+        goto cleanup;
+
+    /**
+     * Loop as long as we have input samples to read or output samples
+     * to write; abort as soon as we have neither.
+     */
+    while (1) {
+        /** Use the encoder's desired frame size for processing. */
+        const int output_frame_size = output_codec_context->frame_size;
+        int finished                = 0;
+
+        /**
+         * Make sure that there is one frame worth of samples in the FIFO
+         * buffer so that the encoder can do its work.
+         * Since the decoder's and the encoder's frame size may differ, we
+         * need to FIFO buffer to store as many frames worth of input samples
+         * that they make up at least one frame worth of output samples.
+         */
+        while (av_audio_fifo_size(fifo) < output_frame_size) {
+            /**
+             * Decode one frame worth of audio samples, convert it to the
+             * output sample format and put it into the FIFO buffer.
+             */
+            if (read_decode_convert_and_store(fifo, input_format_context,
+                                              input_codec_context,
+                                              output_codec_context,
+                                              resample_context, &finished))
+                goto cleanup;
+
+            /**
+             * If we are at the end of the input file, we continue
+             * encoding the remaining audio samples to the output file.
+             */
+            if (finished)
+                break;
+        }
+
+        /**
+         * If we have enough samples for the encoder, we encode them.
+         * At the end of the file, we pass the remaining samples to
+         * the encoder.
+         */
+        while (av_audio_fifo_size(fifo) >= output_frame_size ||
+               (finished && av_audio_fifo_size(fifo) > 0))
+            /**
+             * Take one frame worth of audio samples from the FIFO buffer,
+             * encode it and write it to the output file.
+             */
+            if (load_encode_and_write(fifo, output_format_context,
+                                      output_codec_context))
+                goto cleanup;
+
+        /**
+         * If we are at the end of the input file and have encoded
+         * all remaining samples, we can exit this loop and finish.
+         */
+        if (finished) {
+            int data_written;
+            /** Flush the encoder as it may have delayed frames. */
+            do {
+                if (encode_audio_frame(NULL, output_format_context,
+                                       output_codec_context, &data_written))
+                    goto cleanup;
+            } while (data_written);
+            break;
+        }
+    }
+
+    /** Write the trailer of the output file container. */
+    if (write_output_file_trailer(output_format_context))
+        goto cleanup;
+    ret = 0;
+
+cleanup:
+    if (fifo)
+        av_audio_fifo_free(fifo);
+    swr_free(&resample_context);
+    if (output_codec_context)
+        avcodec_close(output_codec_context);
+    if (output_format_context) {
+        avio_close(output_format_context->pb);
+        avformat_free_context(output_format_context);
+    }
+    if (input_codec_context)
+        avcodec_close(input_codec_context);
+    if (input_format_context)
+        avformat_close_input(&input_format_context);
+
+    return ret;
+}
@@ -105,7 +105,7 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
 Then you may run:

@example
-  ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example

 Notice that @samp{%d} is replaced by the image number.
@@ -118,7 +118,7 @@ the sequence. This is useful if your sequence does not start with
 example will start with @file{img100.jpg}:

@example
-  ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
+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
@@ -128,7 +128,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.

@example
-  x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example

 If you want to sequence them by oldest modified first, substitute
@@ -137,7 +137,7 @@ If you want to sequence them by oldest modified first, substitute
 Then run:

@example
-  ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
+ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example

 The same logic is used for any image format that ffmpeg reads.
@@ -145,7 +145,7 @@ 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
+cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example

@section How do I encode movie to single pictures?
@@ -153,7 +153,7 @@ You can also use @command{cat} to pipe images to ffmpeg:
 Use:

@example
-  ffmpeg -i movie.mpg movie%d.jpg
+ffmpeg -i movie.mpg movie%d.jpg
@end example

 The @file{movie.mpg} used as input will be converted to
@@ -169,7 +169,7 @@ to force the encoding.

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

 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -227,15 +227,15 @@ then you may use any file that DirectShow can read as input.

 Just create an "input.avs" text file with this single line ...
@example
-  DirectShowSource("C:\path to your file\yourfile.asf")
+DirectShowSource("C:\path to your file\yourfile.asf")
@end example
 ... and then feed that text file to ffmpeg:
@example
-  ffmpeg -i input.avs
+ffmpeg -i input.avs
@end example

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

@section How can I join video files?

@@ -368,42 +368,22 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
@end example

-@section -profile option fails when encoding H.264 video with AAC audio
-
-@command{ffmpeg} prints an error like
-
-@example
-Undefined constant or missing '(' in 'baseline'
-Unable to parse option value "baseline"
-Error setting option profile to value baseline.
-@end example
-
-Short answer: write @option{-profile:v} instead of @option{-profile}.
-
-Long answer: this happens because the @option{-profile} option can apply to both
-video and audio.  Specifically the AAC encoder also defines some profiles, none
-of which are named @var{baseline}.
-
-The solution is to apply the @option{-profile} option to the video stream only
-by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
-Appending @code{:v} to it will do exactly that.
-
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.

 Use @option{-dumpgraph -} to find out exactly where the channel layout is
 lost.

-Most likely, it is through @code{auto-inserted aconvert}. Try to understand
+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?
@@ -475,9 +455,10 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.

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

-The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
-information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
-you need the debug information, use the *_g versions.
+The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
+contain full debug information. Those binaries are stripped to create
+@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
+the *_g versions.

@section I do not like the LGPL, can I contribute code under the GPL instead?

@@ -497,7 +478,7 @@ An easy way to get the full list of required libraries in dependency order
 is to use @code{pkg-config}.

@example
-  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example

 See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
@@ -521,10 +502,6 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
 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?
-
-see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
-
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?

 see @url{http://www.ffmpeg.org/~michael/}
@@ -537,11 +514,12 @@ In this specific case please look at RFC 4629 to see how it should be done.

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

-r_frame_rate is NOT the average frame rate, it is the smallest frame rate
+@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then r_frame_rate
-will be 150.
+For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
+will be 150 (it is the least common multiple).
+If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.

@section Why is @code{make fate} not running all tests?

@@ -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
@@ -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
@@ -148,20 +153,20 @@ the synchronisation of the samples directory.

@table @option
@item fate-rsync
-    Download/synchronize sample files to the configured samples directory.
+Download/synchronize sample files to the configured samples directory.

@item fate-list
-    Will list all fate/regression test targets.
+Will list all fate/regression test targets.

@item fate
-    Run the FATE test suite (requires the fate-suite dataset).
+Run the FATE test suite (requires the fate-suite dataset).
@end table

@section Makefile variables

@table @option
@item V
-    Verbosity level, can be set to 0, 1 or 2.
+Verbosity level, can be set to 0, 1 or 2.
    @itemize
        @item 0: show just the test arguments
        @item 1: show just the command used in the test
@@ -169,22 +174,28 @@ the synchronisation of the samples directory.
    @end itemize

@item SAMPLES
-    Specify or override the path to the FATE samples at make time, it has a
-    meaning only while running the regression tests.
+Specify or override the path to the FATE samples at make time, it has a
+meaning only while running the regression tests.

@item THREADS
-    Specify how many threads to use while running regression tests, it is
-    quite useful to detect thread-related regressions.
+Specify how many threads to use while running regression tests, it is
+quite useful to detect thread-related regressions.
+
@item THREAD_TYPE
-    Specify which threading strategy test, either @var{slice} or @var{frame},
-    by default @var{slice+frame}
+Specify which threading strategy test, either @var{slice} or @var{frame},
+by default @var{slice+frame}
+
@item CPUFLAGS
-    Specify CPU flags.
+Specify CPU flags.
+
@item TARGET_EXEC
-    Specify or override the wrapper used to run the tests.
-    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
-    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-    through @command{ssh}.
+Specify or override the wrapper used to run the tests.
+The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
+@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
+through @command{ssh}.
+
+@item GEN
+Set to @var{1} to generate the missing or mismatched references.
@end table

@section Examples
@@ -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=
@@ -17,27 +17,7 @@ libavdevice library.

@c man end DESCRIPTION

-@chapter Device Options
-@c man begin DEVICE OPTIONS
-
-The libavdevice library provides the same interface as
-libavformat. Namely, an input device is considered like a demuxer, and
-an output device like a muxer, and the interface and generic device
-options are the same provided by libavformat (see the ffmpeg-formats
-manual).
-
-In addition each input or output device may support so-called private
-options, which are specific for that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the device
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-@c man end DEVICE OPTIONS
-
-@include indevs.texi
-@include outdevs.texi
+@include devices.texi

@chapter See Also

@@ -17,138 +17,7 @@ provided by the libavformat library.

@c man end DESCRIPTION

-@chapter Format Options
-@c man begin FORMAT OPTIONS
-
-The libavformat library provides some generic global options, which
-can be set on all the muxers and demuxers. In addition each muxer or
-demuxer may support so-called private options, which are specific for
-that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-The list of supported options follows:
-
-@table @option
-@item avioflags @var{flags} (@emph{input/output})
-Possible values:
-@table @samp
-@item direct
-Reduce buffering.
-@end table
-
-@item probesize @var{integer} (@emph{input})
-Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will allow to detect more
-information in case it is dispersed into the stream, but will increase
-latency. Must be an integer not lesser than 32. It is 5000000 by default.
-
-@item packetsize @var{integer} (@emph{output})
-Set packet size.
-
-@item fflags @var{flags} (@emph{input/output})
-Set format flags.
-
-Possible values:
-@table @samp
-@item ignidx
-Ignore index.
-@item genpts
-Generate PTS.
-@item nofillin
-Do not fill in missing values that can be exactly calculated.
-@item noparse
-Disable AVParsers, this needs @code{+nofillin} too.
-@item igndts
-Ignore DTS.
-@item discardcorrupt
-Discard corrupted frames.
-@item sortdts
-Try to interleave output packets by DTS.
-@item keepside
-Do not merge side data.
-@item latm
-Enable RTP MP4A-LATM payload.
-@item nobuffer
-Reduce the latency introduced by optional buffering
-@end table
-
-@item analyzeduration @var{integer} (@emph{input})
-Specify how many microseconds are analyzed to 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 positive. 1 enables, 0 disables, default
-of -1 enables when required by target format.
-@end table
-
-@c man end FORMAT OPTIONS
-
-@include demuxers.texi
-@include muxers.texi
-@include metadata.texi
+@include formats.texi

@chapter See Also

@@ -12,235 +12,14 @@
@chapter Description
@c man begin DESCRIPTION

-The FFmpeg resampler provides an high-level interface to the
+The FFmpeg resampler provides a high-level interface to the
 libswresample library audio resampling utilities. In particular it
-allows to perform audio resampling, audio channel layout rematrixing,
+allows one to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.

@c man end DESCRIPTION

-@chapter Resampler Options
-@c man begin RESAMPLER OPTIONS
-
-The audio resampler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, @var{option}=@var{value} for the aresample filter,
-by setting the value explicitly in the
-@code{SwrContext} options or using the @file{libavutil/opt.h} API for
-programmatic use.
-
-@table @option
-
-@item ich, in_channel_count
-Set the number of input channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{in_channel_layout} is set.
-
-@item och, out_channel_count
-Set the number of output channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{out_channel_layout} is set.
-
-@item uch, used_channel_count
-Set the number of used 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.
-
-@end table
-
-@c man end RESAMPLER OPTIONS
+@include resampler.texi

@chapter See Also

@@ -12,111 +12,13 @@
@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
+The FFmpeg rescaler provides a high-level interface to the libswscale
+library image conversion utilities. In particular it allows one to perform
 image rescaling and pixel format conversion.

@c man end DESCRIPTION

-@chapter Scaler Options
-@c man begin SCALER OPTIONS
-
-The video scaler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools. For programmatic use, they can be set explicitly in the
-@code{SwsContext} options or through the @file{libavutil/opt.h} API.
-
-@table @option
-
-@item sws_flags
-Set the scaler flags. This is also used to set the scaling
-algorithm. Only a single algorithm should be selected.
-
-It accepts the following values:
-@table @samp
-@item fast_bilinear
-Select fast bilinear scaling algorithm.
-
-@item bilinear
-Select bilinear scaling algorithm.
-
-@item bicubic
-Select bicubic scaling algorithm.
-
-@item experimental
-Select experimental scaling algorithm.
-
-@item neighbor
-Select nearest neighbor rescaling algorithm.
-
-@item area
-Select averaging area rescaling algorithm.
-
-@item bicubiclin
-Select bicubic scaling algorithm for the luma component, bilinear for
-chroma components.
-
-@item gauss
-Select Gaussian rescaling algorithm.
-
-@item sinc
-Select sinc rescaling algorithm.
-
-@item lanczos
-Select lanczos rescaling algorithm.
-
-@item spline
-Select natural bicubic spline rescaling algorithm.
-
-@item print_info
-Enable printing/debug logging.
-
-@item accurate_rnd
-Enable accurate rounding.
-
-@item full_chroma_int
-Enable full chroma interpolation.
-
-@item full_chroma_inp
-Select full chroma input.
-
-@item bitexact
-Enable bitexact output.
-@end table
-
-@item srcw
-Set source width.
-
-@item srch
-Set source height.
-
-@item dstw
-Set destination width.
-
-@item dsth
-Set destination height.
-
-@item src_format
-Set source pixel format (must be expressed as an integer).
-
-@item dst_format
-Set destination pixel format (must be expressed as an integer).
-
-@item src_range
-Select source range.
-
-@item dst_range
-Select destination range.
-
-@item param0, param1
-Set scaling algorithm parameters. The specified values are specific of
-some scaling algorithms and ignored by others. The specified values
-are floating point number values.
-
-@end table
-
-@c man end SCALER OPTIONS
+@include scaler.texi

@chapter See Also

@@ -17,8 +17,7 @@ by the libavutil library.

@c man end DESCRIPTION

-@include syntax.texi
-@include eval.texi
+@include utils.texi

@chapter See Also

@@ -16,26 +16,26 @@ ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_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
@@ -50,7 +50,7 @@ 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 -bufsize 64k output.avi
@end example
@@ -80,11 +80,22 @@ The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:

@example
- _______              ______________               _________              ______________            ________
-|       |            |              |             |         |            |              |          |        |
-| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
-| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
-|_______|            |______________|             |_________|            |______________|          |________|
+ _______              ______________
+|       |            |              |
+| input |  demuxer   | encoded data |   decoder
+| file  | ---------> | packets      | -----+
+|_______|            |______________|      |
+                                           v
+                                       _________
+                                      |         |
+                                      | decoded |
+                                      | frames  |
+ ________             ______________  |_________|
+|        |           |              |      |
+| output | <-------- | encoded data | <----+
+| file   |   muxer   | packets      |   encoder
+|________|           |______________|
+

@end example

@@ -96,14 +107,14 @@ 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
@@ -112,11 +123,11 @@ the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:

@example
- _________                        __________              ______________
-|         |                      |          |            |              |
-| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
-| frames  | -------------------> | frames   | ---------> | packets      |
-|_________|                      |__________|            |______________|
+ _________               __________              ______________
+|         |  simple     |          |            |              |
+| decoded |  fltrgrph   | filtered |  encoder   | encoded data |
+| frames  | ----------> | frames   | ---------> | packets      |
+|_________|             |__________|            |______________|

@end example

@@ -125,10 +136,10 @@ Simple filtergraphs are configured with the per-stream @option{-filter} option
 A simple filtergraph for video can look for example like this:

@example
- _______        _____________        _______        _____        ________
-|       |      |             |      |       |      |     |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
-|_______|      |_____________|      |_______|      |_____|      |________|
+ _______        _____________        _______        ________
+|       |      |             |      |       |      |        |
+| input | ---> | deinterlace | ---> | scale | ---> | output |
+|_______|      |_____________|      |_______|      |________|

@end example

@@ -139,7 +150,7 @@ 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:

@@ -164,9 +175,11 @@ 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.
@@ -176,7 +189,7 @@ 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
 _______              ______________            ________
@@ -188,7 +201,7 @@ 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
@@ -196,14 +209,14 @@ 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.

@@ -212,7 +225,7 @@ described.
@chapter Options
@c man begin OPTIONS

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

@section Main options

@@ -220,7 +233,7 @@ 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})
@@ -230,7 +243,8 @@ 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})
@@ -269,30 +283,33 @@ Set the file size limit, expressed in bytes.

@item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
-@var{position}. When used as an output option (before an output filename),
-decodes but discards input until the timestamps reach @var{position}. This is
-slower, but more accurate.
+@var{position}. Note the in most formats it is not possible to seek exactly, so
+@command{ffmpeg} will seek to the closest seek point before @var{position}.
+When transcoding and @option{-accurate_seek} is enabled (the default), this
+extra segment between the seek point and @var{position} will be decoded and
+discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
+will be preserved.
+
+When used as an output option (before an output filename), decodes but discards
+input until the timestamps reach @var{position}.

@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.

@item -itsoffset @var{offset} (@emph{input})
-Set the input time offset in seconds.
-@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
-The offset is added to the timestamps of the input files.
-Specifying a positive offset means that the corresponding
-streams are delayed by @var{offset} seconds.
+Set the input time offset.

-@item -timestamp @var{time} (@emph{output})
+@var{offset} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+The offset is added to the timestamps of the input files. Specifying
+a positive offset means that the corresponding streams are delayed by
+the time duration specified in @var{offset}.
+
+@item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.
-The syntax for @var{time} is:
-@example
-now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
-@end example
-If the value is "now" it takes the current time.
-Time is local time unless 'Z' or 'z' is appended, in which case it is
-interpreted as UTC.
-If the year-month-day part is not specified it takes the current
-year-month-day.
+
+@var{date} must be a time duration specification,
+see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.

@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
 Set a metadata key/value pair.
@@ -339,29 +356,40 @@ Stop writing to the stream after @var{framecount} frames.

@item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
@itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
-Use fixed quality scale (VBR). The meaning of @var{q} is
+Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
 codec-dependent.
+If @var{qscale} is used without a @var{stream_specifier} then it applies only
+to the video stream, this is to maintain compatibility with previous behavior
+and as specifying the same codec specific value to 2 different codecs that is
+audio and video generally is not what is intended when no stream_specifier is
+used.

@anchor{filter_option}
-@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream})
-Create the filter graph specified by @var{filter_graph} and use it to
+@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{filter_graph} is a description of the filter graph to apply to
+@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 filter graph, the input is associated
+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 filter graphs with multiple inputs and/or outputs.
+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.

@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}.
@@ -459,6 +487,10 @@ 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.

+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.

@@ -484,11 +516,8 @@ prefix is ``ffmpeg2pass''. The complete file name will be
@file{PREFIX-N.log}, where N is a number specific to the output
 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})
-Create the filter graph specified by @var{filter_graph} and use it to
+@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}.
@@ -504,7 +533,7 @@ 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.

@@ -519,10 +548,6 @@ 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
@@ -601,6 +626,42 @@ 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
 beginning.
+
+@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
+Use hardware acceleration to decode the matching stream(s). The allowed values
+of @var{hwaccel} are:
+@table @option
+@item none
+Do not use any hardware acceleration (the default).
+
+@item auto
+Automatically select the hardware acceleration method.
+
+@item vdpau
+Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
+@end table
+
+This option has no effect if the selected hwaccel is not available or not
+supported by the chosen decoder.
+
+Note that most acceleration methods are intended for playback and will not be
+faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
+will usually need to copy the decoded frames from the GPU memory into the system
+memory, resulting in further performance loss. This option is thus mainly
+useful for testing.
+
+@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
+Select a device to use for hardware acceleration.
+
+This option only makes sense when the @option{-hwaccel} option is also
+specified. Its exact meaning depends on the specific hardware acceleration
+method chosen.
+
+@table @option
+@item vdpau
+For VDPAU, this option specifies the X11 display/screen to use. If this option
+is not specified, the value of the @var{DISPLAY} environment variable is used
+@end table
@end table

@section Audio Options
@@ -628,8 +689,8 @@ Set the audio codec. This is an alias for @code{-codec:a}.
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.

-@item -af @var{filter_graph} (@emph{output})
-Create the filter graph specified by @var{filter_graph} and use it to
+@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}.
@@ -653,8 +714,6 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
@section Subtitle options:

@table @option
-@item -slang @var{code}
-Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{output})
@@ -680,6 +739,9 @@ 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
@@ -859,13 +921,12 @@ Dump each input packet to stderr.
 When dumping packets, also dump the payload.
@item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
+or live input stream (e.g. when reading from a file). Should not be used
+with actual grab devices or live input streams (where it can cause packet
+loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
-of the input(s). It is useful for real-time output (e.g. live streaming). If
-your input(s) is coming from some other live streaming source (through HTTP or
-UDP for example) the server might already be in real-time, thus the option will
-likely not be required. On the other hand, this is meaningful if your input(s)
-is a file you are trying to push in real-time.
+of the input(s). It is useful for real-time output (e.g. live streaming).
@item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@@ -884,7 +945,7 @@ 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.
@@ -896,6 +957,10 @@ 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.
@@ -905,6 +970,11 @@ 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.
+
+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
@@ -913,7 +983,8 @@ 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, the output timestamps may mismatch with the input
+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}
@@ -971,7 +1042,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
@end example

-@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
+@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
 Force a tag/fourcc for matching streams.

@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@@ -983,10 +1054,10 @@ ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg

@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 the ``Filtergraph syntax'' section of the
+the filtergraph, as described in the ``Filtergraph syntax'' section of the
 ffmpeg-filters manual.

 Input link labels must refer to input streams using the
@@ -1028,6 +1099,32 @@ To generate 5 seconds of pure red video using lavfi @code{color} source:
@example
 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 -accurate_seek (@emph{input})
+This option enables or disables accurate seeking in input files with the
+@option{-ss} option. It is enabled by default, so seeking is accurate when
+transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
+e.g. when copying some streams and transcoding the others.
+
+@item -override_ffserver (@emph{global})
+Overrides the input specifications from @command{ffserver}. Using this
+option you can map any input stream to @command{ffserver} and control
+many aspects of the encoding from @command{ffmpeg}. Without this
+option @command{ffmpeg} will transmit to @command{ffserver} what is
+requested by @command{ffserver}.
+
+The option is intended for cases where features are needed that cannot be
+specified to @command{ffserver} but can be to @command{ffmpeg}.
+
@end table

 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1085,7 +1182,7 @@ then it will search for the file @file{libvpx-1080p.ffpreset}.

@itemize
@item
-For streaming at very low bitrate application, use a low frame rate
+For streaming at very low bitrates, use a low frame rate
 and a small GOP size. This is especially true for RealVideo where
 the Linux player does not seem to be very fast, so it can miss
 frames. An example is:
@@ -1164,14 +1261,14 @@ standard mixer.
 Grab the X11 display with ffmpeg via

@example
-ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
+ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as
 the DISPLAY environment variable.

@example
-ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
+ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as the DISPLAY environment
@@ -1308,11 +1405,11 @@ 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 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
@end example

-The resulting output file @file{test12.avi} will contain first four streams from
-the input file in reverse order.
+The resulting output file @file{test12.nut} will contain the first four streams
+from the input files in reverse order.

@item
 To force CBR video output:
@@ -1330,9 +1427,42 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end itemize
@c man end EXAMPLES

+@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{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},
@@ -1346,6 +1476,12 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@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),
@@ -24,7 +24,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS

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

@section Main options

@@ -73,17 +73,23 @@ 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}
-Create the filter graph specified by @var{filter_graph} and use it to
+@item -vf @var{filtergraph}
+Create the filtergraph specified by @var{filtergraph} and use it to
 filter the video stream.

-@var{filter_graph} is a description of the filter graph to apply to
+@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 filter graph, the input is associated to the label
+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
+sources and sinks).
+
@item -i @var{input_file}
 Read @var{input_file}.
@end table
@@ -93,9 +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
@@ -164,17 +174,26 @@ Toggle full screen.
 Pause.

@item a
-Cycle audio channel.
+Cycle audio channel in the curret program.

@item v
 Cycle video channel.

@item t
-Cycle subtitle channel.
+Cycle subtitle channel in the current program.
+
+@item c
+Cycle program.

@item w
 Show audio waves.

+@item s
+Step to the next frame.
+
+Pause if the stream is not already paused, step to the next video
+frame, and pause.
+
@item left/right
 Seek backward/forward 10 seconds.

@@ -182,6 +201,8 @@ Seek backward/forward 10 seconds.
 Seek backward/forward 1 minute.

@item page down/page up
+Seek to the previous/next chapter.
+or if there are no chapters
 Seek backward/forward 10 minutes.

@item mouse click
@@ -191,9 +212,43 @@ Seek to percentage in file corresponding to fraction of width.

@c man end

+@include config.texi
+@ifset config-all
+@set config-readonly
+@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},
@@ -207,6 +262,12 @@ Seek to percentage in file corresponding to fraction of width.
@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),
@@ -44,14 +44,15 @@ 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.
+and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
+section.

@c man end

@chapter Options
@c man begin OPTIONS

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

@section Main options

@@ -112,7 +113,7 @@ ffprobe -show_packets -select_streams v:1 INPUT
@end example

@item -show_data
-Show payload data, as an hexadecimal and ASCII dump. Coupled with
+Show payload data, as a hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.

@@ -196,11 +197,11 @@ The information for each single packet is printed within a dedicated
 section with name "PACKET".

@item -show_frames
-Show information about each frame contained in the input multimedia
-stream.
+Show information about each frame and subtitle contained in the input
+multimedia stream.

 The information for each single frame is printed within a dedicated
-section with name "FRAME".
+section with name "FRAME" or "SUBTITLE".

@item -show_streams
 Show information about each media stream contained in the input
@@ -209,6 +210,18 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".

+@item -show_programs
+Show information about programs and their streams contained in the input
+multimedia stream.
+
+Each media stream information is printed within a dedicated section
+with name "PROGRAM_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.
@@ -217,6 +230,70 @@ corresponding stream section.
 Count the number of packets per stream and report it in the
 corresponding stream section.

+@item -read_intervals @var{read_intervals}
+
+Read only the specified intervals. @var{read_intervals} must be a
+sequence of interval specifications separated by ",".
+@command{ffprobe} will seek to the interval starting point, and will
+continue reading from that.
+
+Each interval is specified by two optional parts, separated by "%".
+
+The first part specifies the interval start position. It is
+interpreted as an abolute position, or as a relative offset from the
+current position if it is preceded by the "+" character. If this first
+part is not specified, no seeking will be performed when reading this
+interval.
+
+The second part specifies the interval end position. It is interpreted
+as an absolute position, or as a relative offset from the current
+position if it is preceded by the "+" character. If the offset
+specification starts with "#", it is interpreted as the number of
+packets to read (not including the flushing packets) from the interval
+start. If no second part is specified, the program will read until the
+end of the input.
+
+Note that seeking is not accurate, thus the actual interval start
+point may be different from the specified position. Also, when an
+interval duration is specified, the absolute end time will be computed
+by adding the duration to the interval start point found by seeking
+the file, rather than to the specified start value.
+
+The formal syntax is given by:
+@example
+@var{INTERVAL}  ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
+@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
+@end example
+
+A few examples follow.
+@itemize
+@item
+Seek to time 10, read packets until 20 seconds after the found seek
+point, then seek to position @code{01:30} (1 minute and thirty
+seconds) and read packets until position @code{01:45}.
+@example
+10%+20,01:30%01:45
+@end example
+
+@item
+Read only 42 packets after seeking to position @code{01:23}:
+@example
+01:23%+#42
+@end example
+
+@item
+Read only the first 20 seconds from the start:
+@example
+%+20
+@end example
+
+@item
+Read from the start until position @code{02:30}:
+@example
+%02:30
+@end example
+@end itemize
+
@item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -260,6 +337,39 @@ 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 ":".

+All writers support the following options:
+
+@table @option
+@item string_validation, sv
+Set string validation mode.
+
+The following values are accepted.
+@table @samp
+@item fail
+The writer will fail immediately in case an invalid string (UTF-8)
+sequence or code point is found in the input. This is especially
+useful to validate input metadata.
+
+@item ignore
+Any validation error will be ignored. This will result in possibly
+broken output, especially with the json or xml writer.
+
+@item replace
+The writer will substitute invalid UTF-8 sequences or code points with
+the string specified with the @option{string_validation_replacement}.
+@end table
+
+Default value is @samp{replace}.
+
+@item string_validation_replacement, svr
+Set replacement string to use in case @option{string_validation} is
+set to @samp{replace}.
+
+In case the option is not specified, the writer will assume the empty
+string, that is it will remove the invalid sequences from the input
+strings.
+@end table
+
 A description of the currently available writers follows.

@section default
@@ -274,8 +384,8 @@ keyN=valN
 [/SECTION]
@end example

-Metadata tags are printed as a line in the corresponding FORMAT or
-STREAM section, and are prefixed by the string "TAG:".
+Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
+PROGRAM_STREAM section, and are prefixed by the string "TAG:".

 A description of the accepted options follows.

@@ -487,10 +597,44 @@ DV, GXF and AVI timecodes are available in format metadata
@end itemize
@c man end TIMECODE

+@include config.texi
+@ifset config-all
+@set config-readonly
+@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
-@url{ffplay.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@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},
@@ -503,6 +647,12 @@ DV, GXF and AVI timecodes are available in format metadata
@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),
@@ -11,6 +11,8 @@
            <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="programs" type="ffprobe:programsType" 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" />
@@ -26,7 +28,10 @@

    <xsd:complexType name="framesType">
        <xsd:sequence>
-            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
        </xsd:sequence>
    </xsd:complexType>

@@ -56,6 +61,8 @@
      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
      <xsd:attribute name="pkt_dts"       type="xsd:long" />
      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
+      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
+      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
      <xsd:attribute name="pkt_duration"  type="xsd:long" />
      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
      <xsd:attribute name="pkt_pos"       type="xsd:long" />
@@ -78,7 +85,16 @@
      <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="subtitleType">
+      <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
+      <xsd:attribute name="pts"                type="xsd:long" />
+      <xsd:attribute name="pts_time"           type="xsd:float"/>
+      <xsd:attribute name="format"             type="xsd:int"  />
+      <xsd:attribute name="start_display_time" type="xsd:int"  />
+      <xsd:attribute name="end_display_time"   type="xsd:int"  />
+      <xsd:attribute name="num_rects"          type="xsd:int"  />
    </xsd:complexType>

    <xsd:complexType name="streamsType">
@@ -87,6 +103,12 @@
        </xsd:sequence>
    </xsd:complexType>

+    <xsd:complexType name="programsType">
+        <xsd:sequence>
+            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+
    <xsd:complexType name="streamDispositionType">
      <xsd:attribute name="default"          type="xsd:int" use="required" />
      <xsd:attribute name="dub"              type="xsd:int" use="required" />
@@ -103,8 +125,8 @@

    <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:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
      </xsd:sequence>

      <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -131,6 +153,7 @@
      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
      <xsd:attribute name="sample_rate"      type="xsd:int"/>
      <xsd:attribute name="channels"         type="xsd:int"/>
+      <xsd:attribute name="channel_layout"   type="xsd:string"/>
      <xsd:attribute name="bits_per_sample"  type="xsd:int"/>

      <xsd:attribute name="id"               type="xsd:string"/>
@@ -147,6 +170,23 @@
      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
    </xsd:complexType>

+    <xsd:complexType name="programType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
+      </xsd:sequence>
+
+      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
+      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="start_time"           type="xsd:float"/>
+      <xsd:attribute name="start_pts"            type="xsd:long"/>
+      <xsd:attribute name="end_time"             type="xsd:float"/>
+      <xsd:attribute name="end_pts"              type="xsd:long"/>
+      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
+      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
+    </xsd:complexType>
+
    <xsd:complexType name="formatType">
      <xsd:sequence>
        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -154,12 +194,14 @@

      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
      <xsd:attribute name="format_long_name" type="xsd:string"/>
      <xsd:attribute name="start_time"       type="xsd:float"/>
      <xsd:attribute name="duration"         type="xsd:float"/>
      <xsd:attribute name="size"             type="xsd:long"/>
      <xsd:attribute name="bit_rate"         type="xsd:long"/>
+      <xsd:attribute name="probe_score"      type="xsd:int"/>
    </xsd:complexType>

    <xsd:complexType name="tagType">
@@ -182,6 +224,25 @@
      <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"/>
@@ -235,7 +235,7 @@ StartSendOnKey

 #<Stream test.ogg>
 #Feed feed1.ffm
-#Title "Stream title"
+#Metadata title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -280,10 +280,10 @@ StartSendOnKey
 #<Stream file.asf>
 #File "/usr/local/httpd/htdocs/test.asf"
 #NoAudio
-#Author "Me"
-#Copyright "Super MegaCorp"
-#Title "Test stream from disk"
-#Comment "Test comment"
+#Metadata author "Me"
+#Metadata copyright "Super MegaCorp"
+#Metadata title "Test stream from disk"
+#Metadata comment "Test comment"
 #</Stream>


@@ -16,11 +16,14 @@ ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION

-@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}).
+@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.
+
+@command{ffserver} is configured through a configuration file, which
+is read at startup. If not explicitly specified, it will read from
+@file{/etc/ffserver.conf}.

@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
@@ -39,10 +42,126 @@ For each feed you can have different output streams in various
 formats, each one specified by a @code{<Stream>} section in the
 configuration file.

+@chapter Detailed description
+
+@command{ffserver} works by forwarding streams encoded by
+@command{ffmpeg}, or pre-recorded streams which are read from disk.
+
+Precisely, @command{ffserver} acts as an HTTP server, accepting POST
+requests from @command{ffmpeg} to acquire the stream to publish, and
+serving RTSP clients or HTTP clients GET requests with the stream
+media content.
+
+A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
+a port where @command{ffserver} is listening.
+
+Each feed is identified by a unique name, corresponding to the name
+of the resource published on @command{ffserver}, and is configured by
+a dedicated @code{Feed} section in the configuration file.
+
+The feed publish URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
+@end example
+
+where @var{ffserver_ip_address} is the IP address of the machine where
+@command{ffserver} is installed, @var{http_port} is the port number of
+the HTTP server (configured through the @option{Port} option), and
+@var{feed_name} is the name of the corresponding feed defined in the
+configuration file.
+
+Each feed is associated to a file which is stored on disk. This stored
+file is used to allow to send pre-recorded data to a player as fast as
+possible when new content is added in real-time to the stream.
+
+A "live-stream" or "stream" is a resource published by
+@command{ffserver}, and made accessible through the HTTP protocol to
+clients.
+
+A stream can be connected to a feed, or to a file. In the first case,
+the published stream is forwarded from the corresponding feed
+generated by a running instance of @command{ffmpeg}, in the second
+case the stream is read from a pre-recorded file.
+
+Each stream is identified by a unique name, corresponding to the name
+of the resource served by @command{ffserver}, and is configured by
+a dedicated @code{Stream} section in the configuration file.
+
+The stream access HTTP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
+@end example
+
+The stream access RTSP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
+@end example
+
+@var{stream_name} is the name of the corresponding stream defined in
+the configuration file. @var{options} is a list of options specified
+after the URL which affects how the stream is served by
+@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
+and RTSP ports configured with the options @var{Port} and
+@var{RTSPPort} respectively.
+
+In case the stream is associated to a feed, the encoding parameters
+must be configured in the stream configuration. They are sent to
+@command{ffmpeg} when setting up the encoding. This allows
+@command{ffserver} to define the encoding parameters used by
+the @command{ffmpeg} encoders.
+
+The @command{ffmpeg} @option{override_ffserver} commandline option
+allows one to override the encoding parameters set by the server.
+
+Multiple streams can be connected to the same feed.
+
+For example, you can have a situation described by the following
+graph:
+@example
+               _________       __________
+              |         |     |          |
+ffmpeg 1 -----| feed 1  |-----| stream 1 |
+    \         |_________|\    |__________|
+     \                    \
+      \                    \   __________
+       \                    \ |          |
+        \                    \| stream 2 |
+         \                    |__________|
+          \
+           \   _________       __________
+            \ |         |     |          |
+             \| feed 2  |-----| stream 3 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+ffmpeg 2 -----| feed 3  |-----| stream 4 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+              | file 1  |-----| stream 5 |
+              |_________|     |__________|
+@end example
+
+@anchor{FFM}
+@section FFM, FFM2 formats
+
+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.
+
@section Status stream

-ffserver supports an HTTP interface which exposes the current status
-of the server.
+@command{ffserver} supports an HTTP interface which exposes the
+current status of the server.

 Simply point your browser to the address of the special status stream
 specified in the configuration file.
@@ -61,27 +180,8 @@ ACL allow 192.168.0.0 192.168.255.255
 then the server will post a page with the status information when
 the special stream @file{status.html} is requested.

-@section What can this do?
-
-When properly configured and running, you can capture video and audio in real
-time from a suitable capture card, and stream it out over the Internet to
-either Windows Media Player or RealAudio player (with some restrictions).
-
-It can also stream from files, though that is currently broken. Very often, a
-web server can be used to serve up the files just as well.
-
-It can stream prerecorded video from .ffm files, though it is somewhat tricky
-to make it work correctly.
-
@section How do I make it work?

-First, build the kit. It *really* helps to have installed LAME first. Then when
-you run the ffserver ./configure, make sure that you have the
-@code{--enable-libmp3lame} flag turned on.
-
-LAME is important as it allows for streaming audio to Windows Media Player.
-Don't ask why the other audio types do not work.
-
 As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:

@@ -109,35 +209,6 @@ You should edit the ffserver.conf file to suit your needs (in terms of
 frame rates etc). Then install ffserver and ffmpeg, write a script to start
 them up, and off you go.

-@section Troubleshooting
-
-@subsection I don't hear any audio, but video is fine.
-
-Maybe you didn't install LAME, or got your ./configure statement wrong. Check
-the ffmpeg output to see if a line referring to MP3 is present. If not, then
-your configuration was incorrect. If it is, then maybe your wiring is not
-set up correctly. Maybe the sound card is not getting data from the right
-input source. Maybe you have a really awful audio interface (like I do)
-that only captures in stereo and also requires that one channel be flipped.
-If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
-starting ffmpeg.
-
-@subsection The audio and video lose sync after a while.
-
-Yes, they do.
-
-@subsection After a long while, the video update rate goes way down in WMP.
-
-Yes, it does. Who knows why?
-
-@subsection WMP 6.4 behaves differently to WMP 7.
-
-Yes, it does. Any thoughts on this would be gratefully received. These
-differences extend to embedding WMP into a web page. [There are two
-object IDs that you can use: The old one, which does not play well, and
-the new one, which does (both tested on the same system). However,
-I suspect that the new one is not available unless you have installed WMP 7].
-
@section What else can it do?

 You can replay video from .ffm files that was recorded earlier.
@@ -177,9 +248,6 @@ specify a time. In addition, ffserver will skip frames until a key_frame
 is found. This further reduces the startup delay by not transferring data
 that will be discarded.

-* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
-the amount of bandwidth consumed by live streams.
-
@section Why does the ?buffer / Preroll stop working after a time?

 It turns out that (on my machine at least) the number of frames successfully
@@ -213,43 +281,588 @@ 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

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

@section Main options

@table @option
@item -f @var{configfile}
-Use @file{configfile} instead of @file{/etc/ffserver.conf}.
+Read configuration file @file{configfile}. If not specified it will
+read by default from @file{/etc/ffserver.conf}.
+
@item -n
-Enable no-launch mode. This option disables all the Launch directives
-within the various <Stream> sections. Since ffserver will not launch
-any ffmpeg instances, you will have to launch them manually.
+Enable no-launch mode. This option disables all the @code{Launch}
+directives within the various @code{<Feed>} sections. Since
+@command{ffserver} will not launch any @command{ffmpeg} instances, you
+will have to launch them manually.
+
@item -d
-Enable debug mode. This option increases log verbosity, directs log
-messages to stdout.
+Enable debug mode. This option increases log verbosity, and directs
+log messages to stdout. When specified, the @option{CustomLog} option
+is ignored.
@end table
+
+@chapter Configuration file syntax
+
+@command{ffserver} reads a configuration file containing global
+options and settings for each stream and feed.
+
+The configuration file consists of global options and dedicated
+sections, which must be introduced by "<@var{SECTION_NAME}
+@var{ARGS}>" on a separate line and must be terminated by a line in
+the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
+
+Currently the following sections are recognized: @samp{Feed},
+@samp{Stream}, @samp{Redirect}.
+
+A line starting with @code{#} is ignored and treated as a comment.
+
+Name of options and sections are case-insensitive.
+
+@section ACL syntax
+An ACL (Access Control List) specifies the address which are allowed
+to access a given stream, or to write a given feed.
+
+It accepts the folling forms
+@itemize
+@item
+Allow/deny access to @var{address}.
+@example
+ACL ALLOW <address>
+ACL DENY <address>
+@end example
+
+@item
+Allow/deny access to ranges of addresses from @var{first_address} to
+@var{last_address}.
+@example
+ACL ALLOW <first_address> <last_address>
+ACL DENY <first_address> <last_address>
+@end example
+@end itemize
+
+You can repeat the ACL allow/deny as often as you like. It is on a per
+stream basis. The first match defines the action. If there are no matches,
+then the default is the inverse of the last ACL statement.
+
+Thus 'ACL allow localhost' only allows access from localhost.
+'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
+allow everybody else.
+
+@section Global options
+@table @option
+@item Port @var{port_number}
+@item RTSPPort @var{port_number}
+
+Set TCP port number on which the HTTP/RTSP server is listening. You
+must select a different port from your standard HTTP web server if it
+is running on the same computer.
+
+If not specified, no corresponding server will be created.
+
+@item BindAddress @var{ip_address}
+@item RTSPBindAddress @var{ip_address}
+Set address on which the HTTP/RTSP server is bound. Only useful if you
+have several network interfaces.
+
+@item MaxHTTPConnections @var{n}
+Set number of simultaneous HTTP connections that can be handled. It
+has to be defined @emph{before} the @option{MaxClients} parameter,
+since it defines the @option{MaxClients} maximum limit.
+
+Default value is 2000.
+
+@item MaxClients @var{n}
+Set number of simultaneous requests that can be handled. Since
+@command{ffserver} is very fast, it is more likely that you will want
+to leave this high and use @option{MaxBandwidth}.
+
+Default value is 5.
+
+@item MaxBandwidth @var{kbps}
+Set the maximum amount of kbit/sec that you are prepared to consume
+when streaming to clients.
+
+Default value is 1000.
+
+@item CustomLog @var{filename}
+Set access log file (uses standard Apache log file format). '-' is the
+standard output.
+
+If not specified @command{ffserver} will produce no log.
+
+In case the commandline option @option{-d} is specified this option is
+ignored, and the log is written to standard output.
+
+@item NoDaemon
+Set no-daemon mode. This option is currently ignored since now
+@command{ffserver} will always work in no-daemon mode, and is
+deprecated.
+@end table
+
+@section Feed section
+
+A Feed section defines a feed provided to @command{ffserver}.
+
+Each live feed contains one video and/or audio sequence coming from an
+@command{ffmpeg} encoder or another @command{ffserver}. This sequence
+may be encoded simultaneously with several codecs at several
+resolutions.
+
+A feed instance specification is introduced by a line in the form:
+@example
+<Feed FEED_FILENAME>
+@end example
+
+where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
+
+The following options are recognized within a Feed section.
+
+@table @option
+@item File @var{filename}
+@item ReadOnlyFile @var{filename}
+Set the path where the feed file is stored on disk.
+
+If not specified, the @file{/tmp/FEED.ffm} is assumed, where
+@var{FEED} is the feed name.
+
+If @option{ReadOnlyFile} is used the file is marked as read-only and
+it will not be deleted or updated.
+
+@item Truncate
+Truncate the feed file, rather than appending to it. By default
+@command{ffserver} will append data to the file, until the maximum
+file size value is reached (see @option{FileMaxSize} option).
+
+@item FileMaxSize @var{size}
+Set maximum size of the feed file in bytes. 0 means unlimited. The
+postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
+recognized.
+
+Default value is 5M.
+
+@item Launch @var{args}
+Launch an @command{ffmpeg} command when creating @command{ffserver}.
+
+@var{args} must be a sequence of arguments to be provided to an
+@command{ffmpeg} instance. The first provided argument is ignored, and
+it is replaced by a path with the same dirname of the @command{ffserver}
+instance, followed by the remaining argument and terminated with a
+path corresponding to the feed.
+
+When the launched process exits, @command{ffserver} will launch
+another program instance.
+
+In case you need a more complex @command{ffmpeg} configuration,
+e.g. if you need to generate multiple FFM feeds with a single
+@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
+
+This option is ignored in case the commandline option @option{-n} is
+specified.
+
+@item ACL @var{spec}
+Specify the list of IP address which are allowed or denied to write
+the feed. Multiple ACL options can be specified.
+@end table
+
+@section Stream section
+
+A Stream section defines a stream provided by @command{ffserver}, and
+identified by a single name.
+
+The stream is sent when answering a request containing the stream
+name.
+
+A stream section must be introduced by the line:
+@example
+<Stream STREAM_NAME>
+@end example
+
+where @var{STREAM_NAME} specifies the unique name of the stream.
+
+The following options are recognized within a Stream section.
+
+Encoding options are marked with the @emph{encoding} tag, and they are
+used to set the encoding parameters, and are mapped to libavcodec
+encoding options. Not all encoding options are supported, in
+particular it is not possible to set encoder private options. In order
+to override the encoding options specified by @command{ffserver}, you
+can use the @command{ffmpeg} @option{override_ffserver} commandline
+option.
+
+Only one of the @option{Feed} and @option{File} options should be set.
+
+@table @option
+@item Feed @var{feed_name}
+Set the input feed. @var{feed_name} must correspond to an existing
+feed defined in a @code{Feed} section.
+
+When this option is set, encoding options are used to setup the
+encoding operated by the remote @command{ffmpeg} process.
+
+@item File @var{filename}
+Set the filename of the pre-recorded input file to stream.
+
+When this option is set, encoding options are ignored and the input
+file content is re-streamed as is.
+
+@item Format @var{format_name}
+Set the format of the output stream.
+
+Must be the name of a format recognized by FFmpeg. If set to
+@samp{status}, it is treated as a status stream.
+
+@item InputFormat @var{format_name}
+Set input format. If not specified, it is automatically guessed.
+
+@item Preroll @var{n}
+Set this to the number of seconds backwards in time to start. Note that
+most players will buffer 5-10 seconds of video, and also you need to allow
+for a keyframe to appear in the data stream.
+
+Default value is 0.
+
+@item StartSendOnKey
+Do not send stream until it gets the first key frame. By default
+@command{ffserver} will send data immediately.
+
+@item MaxTime @var{n}
+Set the number of seconds to run. This value set the maximum duration
+of the stream a client will be able to receive.
+
+A value of 0 means that no limit is set on the stream duration.
+
+@item ACL @var{spec}
+Set ACL for the stream.
+
+@item DynamicACL @var{spec}
+
+@item RTSPOption @var{option}
+
+@item MulticastAddress @var{address}
+
+@item MulticastPort @var{port}
+
+@item MulticastTTL @var{integer}
+
+@item NoLoop
+
+@item FaviconURL @var{url}
+Set favicon (favourite icon) for the server status page. It is ignored
+for regular streams.
+
+@item Author @var{value}
+@item Comment @var{value}
+@item Copyright @var{value}
+@item Title @var{value}
+Set metadata corresponding to the option. All these options are
+deprecated in favor of @option{Metadata}.
+
+@item Metadata @var{key} @var{value}
+Set metadata value on the output stream.
+
+@item NoAudio
+@item NoVideo
+Suppress audio/video.
+
+@item AudioCodec @var{codec_name} (@emph{encoding,audio})
+Set audio codec.
+
+@item AudioBitRate @var{rate} (@emph{encoding,audio})
+Set bitrate for the audio stream in kbits per second.
+
+@item AudioChannels @var{n} (@emph{encoding,audio})
+Set number of audio channels.
+
+@item AudioSampleRate @var{n} (@emph{encoding,audio})
+Set sampling frequency for audio. When using low bitrates, you should
+lower this frequency to 22050 or 11025. The supported frequencies
+depend on the selected audio codec.
+
+@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
+Set generic option for audio stream.
+
+@item AVPresetAudio @var{preset} (@emph{encoding,audio})
+Set preset for audio stream.
+
+@item VideoCodec @var{codec_name} (@emph{encoding,video})
+Set video codec.
+
+@item VideoBitRate @var{n} (@emph{encoding,video})
+Set bitrate for the video stream in kbits per second.
+
+@item VideoBitRateRange @var{range} (@emph{encoding,video})
+Set video bitrate range.
+
+A range must be specified in the form @var{minrate}-@var{maxrate}, and
+specifies the @option{minrate} and @option{maxrate} encoding options
+expressed in kbits per second.
+
+@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
+Set video bitrate tolerance in kbits per second.
+
+@item PixelFormat @var{pixel_format} (@emph{encoding,video})
+Set video pixel format.
+
+@item Debug @var{integer} (@emph{encoding,video})
+Set video @option{debug} encoding option.
+
+@item Strict @var{integer} (@emph{encoding,video})
+Set video @option{strict} encoding option.
+
+@item VideoBufferSize @var{n} (@emph{encoding,video})
+Set ratecontrol buffer size, expressed in KB.
+
+@item VideoFrameRate @var{n} (@emph{encoding,video})
+Set number of video frames per second.
+
+@item VideoSize (@emph{encoding,video})
+Set size of the video frame, must be an abbreviation or in the form
+@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
+in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+Default value is @code{160x128}.
+
+@item VideoIntraOnly (@emph{encoding,video})
+Transmit only intra frames (useful for low bitrates, but kills frame rate).
+
+@item VideoGopSize @var{n} (@emph{encoding,video})
+If non-intra only, an intra frame is transmitted every VideoGopSize
+frames. Video synchronization can only begin at an intra frame.
+
+@item VideoTag @var{tag} (@emph{encoding,video})
+Set video tag.
+
+@item VideoHighQuality (@emph{encoding,video})
+@item Video4MotionVector (@emph{encoding,video})
+
+@item BitExact (@emph{encoding,video})
+Set bitexact encoding flag.
+
+@item IdctSimple (@emph{encoding,video})
+Set simple IDCT algorithm.
+
+@item Qscale @var{n} (@emph{encoding,video})
+Enable constant quality encoding, and set video qscale (quantization
+scale) value, expressed in @var{n} QP units.
+
+@item VideoQMin @var{n} (@emph{encoding,video})
+@item VideoQMax @var{n} (@emph{encoding,video})
+Set video qmin/qmax.
+
+@item VideoQDiff @var{integer} (@emph{encoding,video})
+Set video @option{qdiff} encoding option.
+
+@item LumiMask @var{float} (@emph{encoding,video})
+@item DarkMask @var{float} (@emph{encoding,video})
+Set @option{lumi_mask}/@option{dark_mask} encoding options.
+
+@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
+Set generic option for video stream.
+
+@item AVPresetVideo @var{preset} (@emph{encoding,video})
+Set preset for video stream.
+
+@var{preset} must be the path of a preset file.
+@end table
+
+@subsection Server status stream
+
+A server status stream is a special stream which is used to show
+statistics about the @command{ffserver} operations.
+
+It must be specified setting the option @option{Format} to
+@samp{status}.
+
+@section Redirect section
+
+A redirect section specifies where to redirect the requested URL to
+another page.
+
+A redirect section must be introduced by the line:
+@example
+<Redirect NAME>
+@end example
+
+where @var{NAME} is the name of the page which should be redirected.
+
+It only accepts the option @option{URL}, which specify the redirection
+URL.
+
+@chapter Stream examples
+
+@itemize
+@item
+Multipart JPEG
+@example
+<Stream test.mjpg>
+Feed feed1.ffm
+Format mpjpeg
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Single JPEG
+@example
+<Stream test.jpg>
+Feed feed1.ffm
+Format jpeg
+VideoFrameRate 2
+VideoIntraOnly
+VideoSize 352x240
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Flash
+@example
+<Stream test.swf>
+Feed feed1.ffm
+Format swf
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+</Stream>
+@end example
+
+@item
+ASF compatible
+@example
+<Stream test.asf>
+Feed feed1.ffm
+Format asf
+VideoFrameRate 15
+VideoSize 352x240
+VideoBitRate 256
+VideoBufferSize 40
+VideoGopSize 30
+AudioBitRate 64
+StartSendOnKey
+</Stream>
+@end example
+
+@item
+MP3 audio
+@example
+<Stream test.mp3>
+Feed feed1.ffm
+Format mp2
+AudioCodec mp3
+AudioBitRate 64
+AudioChannels 1
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Ogg Vorbis audio
+@example
+<Stream test.ogg>
+Feed feed1.ffm
+Metadata title "Stream title"
+AudioBitRate 64
+AudioChannels 2
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio only at 32 kbits
+@example
+<Stream test.ra>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio and video at 64 kbits
+@example
+<Stream test.rm>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+VideoBitRate 128
+VideoFrameRate 25
+VideoGopSize 25
+</Stream>
+@end example
+
+@item
+For stream coming from a file: you only need to set the input filename
+and optionally a new format.
+
+@example
+<Stream file.rm>
+File "/usr/local/httpd/htdocs/tlive.rm"
+NoAudio
+</Stream>
+@end example
+
+@example
+<Stream file.asf>
+File "/usr/local/httpd/htdocs/test.asf"
+NoAudio
+Metadata author "Me"
+Metadata copyright "Super MegaCorp"
+Metadata title "Test stream from disk"
+Metadata comment "Test comment"
+</Stream>
+@end example
+@end itemize
+
@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
-The @file{doc/ffserver.conf} example,
+@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},
@@ -263,7 +876,13 @@ The @file{doc/ffserver.conf} example,
@end ifhtml

@ifnothtml
-The @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
+@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)
@@ -1,32 +1,33 @@
-All the numerical options, if not specified otherwise, accept in input
-a string representing a number, which may contain one of the
-SI unit prefixes, for example 'K', 'M', 'G'.
-If 'i' is appended after the prefix, binary prefixes are used,
-which are based on powers of 1024 instead of powers of 1000.
-The 'B' postfix multiplies the value by 8, and can be
-appended after a unit prefix or used alone. This allows using for
-example 'KB', 'MiB', 'G' and 'B' as number postfix.
+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:
@@ -35,20 +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

@@ -57,10 +59,18 @@ Show license.

@item -h, -?, -help, --help [@var{arg}]
 Show help. An optional parameter may be specified to print help about a specific
-item.
+item. If no argument is specified, only basic (non advanced) tool
+options are shown.

 Possible values of @var{arg} are:
@table @option
+@item long
+Print advanced tool options in addition to the basic tool options.
+
+@item full
+Print complete list of options, including shared and private options
+for encoders, decoders, demuxers, muxers, filters, etc.
+
@item decoder=@var{decoder_name}
 Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@@ -77,6 +87,9 @@ 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
@@ -85,14 +98,6 @@ 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.

@@ -123,18 +128,39 @@ Show available sample formats.
@item -layouts
 Show channel names and standard channel layouts.

-@item -loglevel @var{loglevel} | -v @var{loglevel}
+@item -colors
+Show recognized color names.
+
+@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
@@ -168,6 +194,13 @@ to a plain @code{%}
 Errors in parsing the environment variable are not fatal, and will not
 appear in the report.

+@item -hide_banner
+Suppress printing banner.
+
+All FFmpeg tools will normally show a copyright notice, build options
+and library versions. This option can be used to suppress printing
+this information.
+
@item -cpuflags flags (@emph{global})
 Allows setting and clearing cpu flags. This option is intended
 for testing. Do not use it unless you know what you're doing.
@@ -176,7 +209,65 @@ 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_bench
+Benchmark all available OpenCL devices and show the results. This option
+is only available when FFmpeg has been compiled with @code{--enable-opencl}.
+
+@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
@@ -201,11 +292,12 @@ muxer:
 ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example

-All codec AVOptions are obviously per-stream, so the chapter on stream
-specifiers applies to them
+All codec AVOptions are per-stream, and thus a stream specifier
+should be attached to them.

-Note @option{-nooption} syntax cannot be used for boolean AVOptions,
-use @option{-option 0}/@option{-option 1}.
+Note: the @option{-nooption} syntax cannot be used for boolean
+AVOptions, use @option{-option 0}/@option{-option 1}.

-Note2 old undocumented way of specifying per-stream AVOptions by prepending
-v/a/s to the options name is now obsolete and will be removed soon.
+Note: the 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.
@@ -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
 ===========================================
@@ -161,7 +166,7 @@ Buffer references ownership and permissions
      WRITE permission.

    * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and and must request a buffer
+      scale) need the READ permission on input and must request a buffer
      with the WRITE permission.

    * Filters that intend to keep a reference after the filtering process
@@ -199,7 +204,7 @@ Frame scheduling
    filter; these buffered frames must be flushed immediately if a new input
    produces new output.

-    (Example: framerate-doubling filter: filter_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.)

@@ -0,0 +1,212 @@
+@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 seek2any @var{integer} (@emph{input})
+Allow seeking to non-keyframes on demuxer level when supported if set to 1.
+Default is 0.
+
+@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})
+
+Possible values:
+@table @samp
+@item make_non_negative
+Shift timestamps to make them non-negative.
+Also note that this affects only leading negative timestamps, and not
+non-monotonic negative timestamps.
+@item make_zero
+Shift timestamps so that the first timestamp is 0.
+@item auto (default)
+Enables shifting when required by the target format.
+@item disabled
+Disables shifting of timestamp.
+@end table
+
+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.
+
+@item skip_initial_bytes @var{integer} (@emph{input})
+Set number of bytes to skip before reading header and frames if set to 1.
+Default is 0.
+
+@item correct_ts_overflow @var{integer} (@emph{input})
+Correct single timestamp overflows if set to 1. Default is 1.
+
+@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.
+
+@item output_ts_offset @var{offset} (@emph{output})
+Set the output time offset.
+
+@var{offset} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+The offset is added by the muxer to the output timestamps.
+
+Specifying a positive offset means that the corresponding streams are
+delayed bt the time duration specified in @var{offset}. Default value
+is @code{0} (meaning that no offset is applied).
+@end table
+
+@c man end FORMAT OPTIONS
+
+@anchor{Format stream specifiers}
+@section Format stream specifiers
+
+Format stream specifiers allow selection of one or more streams that
+match specific properties.
+
+Possible forms of stream specifiers are:
+@table @option
+@item @var{stream_index}
+Matches the stream with this index.
+
+@item @var{stream_type}[:@var{stream_index}]
+@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 the 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 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 a format-specific ID.
+@end table
+
+The exact semantics of stream specifiers is defined by the
+@code{avformat_match_stream_specifier()} function declared in the
+@file{libavformat/avformat.h} header.
+
+@ifclear config-writeonly
+@include demuxers.texi
+@end ifclear
+@ifclear config-readonly
+@include muxers.texi
+@end ifclear
+@include metadata.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
@@ -89,12 +94,20 @@ Then pass @code{--enable-libtwolame} to configure to enable it.

@section libvpx

-FFmpeg can make use of the libvpx library for VP8 encoding.
+FFmpeg can make use of the libvpx library for VP8/VP9 encoding.

 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.
@@ -109,6 +122,20 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float

+@section x265
+
+FFmpeg can make use of the x265 library for HEVC encoding.
+
+Go to @url{http://x265.org/developers.html} and follow the instructions
+for installing the library. Then pass @code{--enable-libx265} to configure
+to enable it.
+
+@float note
+x265 is under the GNU Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
+details), you must upgrade FFmpeg's license to GPL in order to use it.
+@end float
+
@section libilbc

 iLBC is a narrowband speech codec that has been made freely available
@@ -120,6 +147,41 @@ Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.

+@section libzvbi
+
+libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
+teletext pages and DVB teletext subtitles.
+
+Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
+installing the library. Then pass @code{--enable-libzvbi} to configure to
+enable it.
+
+@float NOTE
+libzvbi is licensed under the GNU General Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
+you must upgrade FFmpeg's license to GPL in order to use it.
+@end float
+
+@section AviSynth
+
+FFmpeg can read AviSynth scripts as input. To enable support, pass
+@code{--enable-avisynth} to configure.  The correct headers are
+included in compat/avisynth/, which allows the user to enable support
+without needing to search for these headers themselves.
+
+For Windows, supported AviSynth variants are
+@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
+@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
+
+For Linux and OS X, the supported AviSynth variant is
+@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
+
+@float NOTE
+AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
+with @code{--enable-avisynth}, and the binaries will work regardless of the
+end user having AviSynth or AvxSynth installed - they'll only need to be
+installed to use AviSynth scripts (obviously).
+@end float


@chapter Supported File Formats, Codecs or Features
@@ -147,13 +209,15 @@ 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 AviSynth                  @tab   @tab X
@item AVR                       @tab   @tab X
    @tab Audio format used on Mac.
@item AVS                       @tab   @tab X
@@ -220,6 +284,8 @@ library:
@item GXF                       @tab X @tab X
    @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
         playout servers.
+@item HNM @tab   @tab X
+    @tab Only version 4 supported, used in some games from Cryo Interactive
@item iCEDraw File              @tab   @tab X
@item ICO                       @tab X @tab X
    @tab Microsoft Windows ICO
@@ -270,6 +336,8 @@ library:
    @tab also known as DVB Transport Stream
@item MPEG-4                    @tab X @tab X
    @tab MPEG-4 is a variant of QuickTime.
+@item Mirillis FIC video        @tab   @tab X
+    @tab No cursor rendering.
@item MIME multipart JPEG       @tab X @tab
@item MSN TCP webcam            @tab   @tab X
    @tab Used by MSN Messenger webcam streams.
@@ -309,6 +377,7 @@ library:
@item raw H.261                 @tab X @tab X
@item raw H.263                 @tab X @tab X
@item raw H.264                 @tab X @tab X
+@item raw HEVC                  @tab X @tab X
@item raw Ingenient MJPEG       @tab   @tab X
@item raw MJPEG                 @tab X @tab X
@item raw MLP                   @tab   @tab X
@@ -322,7 +391,7 @@ library:
@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
@@ -348,11 +417,13 @@ 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
@@ -418,7 +489,6 @@ 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
@@ -458,6 +528,8 @@ following image formats are supported:
    @tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa  @tab X @tab X
    @tab Targa (.TGA) image format
+@item WebP         @tab E @tab X
+    @tab WebP image format, encoding supported through external library libwebp
@item XBM  @tab X @tab X
    @tab X BitMap image format
@item XFace @tab X @tab X
@@ -485,6 +557,7 @@ 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
@@ -567,12 +640,16 @@ 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
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
    @tab encoding supported through external library libx264
-@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration)  @tab  E  @tab  X
+@item HEVC                   @tab  X  @tab  X
+    @tab encoding supported through the external library libx265
+@item HNM version 4          @tab     @tab  X
@item HuffYUV                @tab  X  @tab  X
@item HuffYUV FFmpeg variant @tab  X  @tab  X
@item IBM Ultimotion         @tab     @tab  X
@@ -624,8 +701,6 @@ following image formats are supported:
@item Mobotix MxPEG video    @tab     @tab  X
@item Motion Pixels video    @tab     @tab  X
@item MPEG-1 video           @tab  X  @tab  X
-@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
-@item MPEG-1/2 video (VDPAU acceleration)  @tab     @tab  X
@item MPEG-2 video           @tab  X  @tab  X
@item MPEG-4 part 2          @tab  X  @tab  X
    @tab libxvidcore can be used alternatively for encoding.
@@ -643,6 +718,8 @@ 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 VP9                    @tab  E  @tab  X
+    @tab encoding supported through external library libvpx
@item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
    @tab fourcc: Y216
@item Prores                 @tab     @tab  X
@@ -769,9 +846,11 @@ following image formats are supported:
    @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
@@ -790,8 +869,9 @@ following image formats are supported:
@item Amazing Studio PAF Audio @tab     @tab  X
@item Apple lossless audio   @tab  X  @tab  X
    @tab QuickTime fourcc 'alac'
-@item Atrac 1                @tab     @tab  X
-@item Atrac 3                @tab     @tab  X
+@item ATRAC1                 @tab     @tab  X
+@item ATRAC3                 @tab     @tab  X
+@item ATRAC3+                @tab     @tab  X
@item Bink Audio             @tab     @tab  X
    @tab Used in Bink and Smacker files in many games.
@item CELT                   @tab     @tab  E
@@ -831,7 +911,6 @@ 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.
@@ -886,7 +965,7 @@ 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
@@ -894,7 +973,7 @@ following image formats are supported:
@item Speex                  @tab  E  @tab  E
    @tab supported through external library libspeex
@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
-@item True Audio (TTA)       @tab     @tab  X
+@item 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
@@ -902,7 +981,8 @@ 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 Voxware MetaSound      @tab     @tab  X
+@item WavPack                @tab  X  @tab  X
@item Westwood Audio (SND1)  @tab     @tab  X
@item Windows Media Audio 1  @tab  X  @tab  X
@item Windows Media Audio 2  @tab  X  @tab  X
@@ -925,6 +1005,7 @@ performance on systems without hardware floating point support).
@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 DVB teletext     @tab   @tab X @tab   @tab E
@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
@@ -941,17 +1022,20 @@ performance on systems without hardware floating point support).
@item TED Talks captions @tab @tab X @tab   @tab X
@item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
@item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab   @tab X @tab   @tab X
+@item WebVTT           @tab X @tab X @tab   @tab X
@item XSUB             @tab   @tab   @tab X @tab X
@end multitable

@code{X} means that the feature is supported.

+@code{E} means that support is provided through an external library.
+
@section Network Protocols

@multitable @columnfractions .4 .1
@item Name         @tab Support
@item file         @tab X
+@item FTP          @tab X
@item Gopher       @tab X
@item HLS          @tab X
@item HTTP         @tab X
@@ -967,6 +1051,7 @@ performance on systems without hardware floating point support).
@item RTMPTS       @tab X
@item RTP          @tab X
@item SCTP         @tab X
+@item SFTP         @tab E
@item TCP          @tab X
@item TLS          @tab X
@item UDP          @tab X
@@ -986,15 +1071,16 @@ performance on systems without hardware floating point support).
@item caca              @tab        @tab X
@item DV1394            @tab X      @tab
@item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab
+@item Linux framebuffer @tab X      @tab X
@item JACK              @tab X      @tab
@item LIBCDIO           @tab X
@item LIBDC1394         @tab X      @tab
@item OpenAL            @tab X
+@item OpenGL            @tab        @tab X
@item OSS               @tab X      @tab X
-@item Pulseaudio        @tab X      @tab
+@item PulseAudio        @tab X      @tab X
@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
@@ -299,7 +299,7 @@ the current branch history.
 git commit --amend
@end example

-allows to amend the last commit details quickly.
+allows one to amend the last commit details quickly.

@example
 git rebase -i origin/master
@@ -1,273 +0,0 @@
-
-About Git write access:
-~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-
-Before everything else, you should know how to use GIT properly.
-Luckily Git comes with excellent documentation.
-
-  git --help
-  man git
-
-shows you the available subcommands,
-
-  git <command> --help
-  man git-<command>
-
-shows information about the subcommand <command>.
-
-The most comprehensive manual is the website Git Reference
-
-http://gitref.org/
-
-For more information about the Git project, visit
-
-http://git-scm.com/
-
-Consult these resources whenever you have problems, they are quite exhaustive.
-
-You do not need a special username or password.
-All you need is to provide a ssh public key to the Git server admin.
-
-What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines. Read it at least once, if you are granted commit privileges to the
-FFmpeg project you are expected to be familiar with these rules.
-
-
-
-I. BASICS:
-==========
-
-0. Get GIT:
-
-  Most distributions have a git package, if not
-  You can get git from http://git-scm.com/
-
-
-1. Cloning the source tree:
-
-    git clone git://source.ffmpeg.org/ffmpeg <target>
-
-  This will put the FFmpeg sources into the directory <target>.
-
-    git clone git@source.ffmpeg.org:ffmpeg <target>
-
-  This will put the FFmpeg sources into the directory <target> and let
-  you push back your changes to the remote repository.
-
-
-2. Updating the source tree to the latest revision:
-
-    git pull (--ff-only)
-
-  pulls in the latest changes from the tracked branch. The tracked branch
-  can be remote. By default the master branch tracks the branch master in
-  the remote origin.
-  Caveat: Since merge commits are forbidden at least for the initial
-          months of git --ff-only or --rebase (see below) are recommended.
-          --ff-only will fail and not create merge commits if your branch
-          has diverged (has a different history) from the tracked branch.
-
-2.a Rebasing your local branches:
-
-    git pull --rebase
-
-  fetches the changes from the main repository and replays your local commits
-  over it. This is required to keep all your local changes at the top of
-  FFmpeg's master tree. The master tree will reject pushes with merge commits.
-
-
-3. Adding/removing files/directories:
-
-    git add [-A] <filename/dirname>
-    git rm [-r] <filename/dirname>
-
-  GIT needs to get notified of all changes you make to your working
-  directory that makes files appear or disappear.
-  Line moves across files are automatically tracked.
-
-
-4. Showing modifications:
-
-    git diff <filename(s)>
-
-  will show all local modifications in your working directory as unified diff.
-
-
-5. Inspecting the changelog:
-
-    git log <filename(s)>
-
-  You may also use the graphical tools like gitview or gitk or the web
-  interface available at http://source.ffmpeg.org
-
-6. Checking source tree status:
-
-    git status
-
-  detects all the changes you made and lists what actions will be taken in case
-  of a commit (additions, modifications, deletions, etc.).
-
-
-7. Committing:
-
-    git diff --check
-
-  to double check your changes before committing them to avoid trouble later
-  on. All experienced developers do this on each and every commit, no matter
-  how small.
-  Every one of them has been saved from looking like a fool by this many times.
-  It's very easy for stray debug output or cosmetic modifications to slip in,
-  please avoid problems through this extra level of scrutiny.
-
-  For cosmetics-only commits you should get (almost) empty output from
-
-    git diff -w -b <filename(s)>
-
-  Also check the output of
-
-    git status
-
-  to make sure you don't have untracked files or deletions.
-
-    git add [-i|-p|-A] <filenames/dirnames>
-
-  Make sure you have told git your name and email address, e.g. by running
-    git config --global user.name "My Name"
-    git config --global user.email my@email.invalid
-  (--global to set the global configuration for all your git checkouts).
-
-  Git will select the changes to the files for commit. Optionally you can use
-  the interactive or the patch mode to select hunk by hunk what should be
-  added to the commit.
-
-    git commit
-
-  Git will commit the selected changes to your current local branch.
-
-  You will be prompted for a log message in an editor, which is either
-  set in your personal configuration file through
-
-    git config core.editor
-
-  or set by one of the following environment variables:
-  GIT_EDITOR, VISUAL or EDITOR.
-
-  Log messages should be concise but descriptive. Explain why you made a change,
-  what you did will be obvious from the changes themselves most of the time.
-  Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
-  levels look at and educate themselves while reading through your code. Don't
-  include filenames in log messages, Git provides that information.
-
-  Possibly make the commit message have a terse, descriptive first line, an
-  empty line and then a full description. The first line will be used to name
-  the patch by git format-patch.
-
-
-8. Renaming/moving/copying files or contents of files:
-
-  Git automatically tracks such changes, making those normal commits.
-
-    mv/cp path/file otherpath/otherfile
-
-    git add [-A] .
-
-    git commit
-
-  Do not move, rename or copy files of which you are not the maintainer without
-  discussing it on the mailing list first!
-
-9. Reverting broken commits
-
-    git revert <commit>
-
-  git revert will generate a revert commit. This will not make the faulty
-  commit disappear from the history.
-
-    git reset <commit>
-
-  git reset will uncommit the changes till <commit> rewriting the current
-  branch history.
-
-    git commit --amend
-
-  allows to amend the last commit details quickly.
-
-    git rebase -i origin/master
-
-  will replay local commits over the main repository allowing to edit,
-  merge or remove some of them in the process.
-
-  Note that the reset, commit --amend and rebase rewrite history, so you
-  should use them ONLY on your local or topic branches.
-
-  The main repository will reject those changes.
-
-10. Preparing a patchset.
-
-    git format-patch <commit> [-o directory]
-
-  will generate a set of patches for each commit between <commit> and
-  current HEAD. E.g.
-
-    git format-patch origin/master
-
-  will generate patches for all commits on current branch which are not
-  present in upstream.
-  A useful shortcut is also
-
-    git format-patch -n
-
-  which will generate patches from last n commits.
-  By default the patches are created in the current directory.
-
-11. Sending patches for review
-
-    git send-email <commit list|directory>
-
-  will send the patches created by git format-patch or directly generates
-  them. All the email fields can be configured in the global/local
-  configuration or overridden by command line.
-  Note that this tool must often be installed separately (e.g. git-email
-  package on Debian-based distros).
-
-12. Pushing changes to remote trees
-
-    git push
-
-  Will push the changes to the default remote (origin).
-  Git will prevent you from pushing changes if the local and remote trees are
-  out of sync. Refer to 2 and 2.a to sync the local tree.
-
-    git remote add <name> <url>
-
-  Will add additional remote with a name reference, it is useful if you want
-  to push your local branch for review on a remote host.
-
-    git push <remote> <refspec>
-
-  Will push the changes to the remote repository. Omitting refspec makes git
-  push update all the remote branches matching the local ones.
-
-13. Finding a specific svn revision
-
-  Since version 1.7.1 git supports ':/foo' syntax for specifying commits
-  based on a regular expression. see man gitrevisions
-
-    git show :/'as revision 23456'
-
-  will show the svn changeset r23456. With older git versions searching in
-  the git log output is the easiest option (especially if a pager with
-  search capabilities is used).
-  This commit can be checked out with
-
-    git checkout -b svn_23456 :/'as revision 23456'
-
-  or for git < 1.7.1 with
-
-    git checkout -b svn_23456 $SHA1
-
-  where $SHA1 is the commit SHA1 from the 'git log' output.
-
-
-Contact the project admins <root at ffmpeg dot org> if you have technical
-problems with the GIT server.
@@ -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.
@@ -409,7 +409,7 @@ OpenAL is part of Core Audio, the official Mac OS X Audio interface.
 See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
@end table

-This device allows to capture from an audio input device handled
+This device allows one to capture from an audio input device handled
 through OpenAL.

 You need to specify the name of the device to capture in the provided
@@ -485,87 +485,52 @@ For more information about OSS see:

@section pulse

-pulseaudio input device.
+PulseAudio input device.

-To enable this input device during configuration you need libpulse-simple
-installed in your system.
+To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.

 The filename to provide to the input device is a source device or the
 string "default"

-To list the pulse source devices and their properties you can invoke
+To list the PulseAudio source devices and their properties you can invoke
 the command @command{pactl list sources}.

+More information about PulseAudio can be found on @url{http://www.pulseaudio.org}.
+
+@subsection Options
+@table @option
+@item server
+Connect to a specific PulseAudio server, specified by an IP address.
+Default server is used when not provided.
+
+@item name
+Specify the application name PulseAudio will use when showing active clients,
+by default it is the @code{LIBAVFORMAT_IDENT} string.
+
+@item stream_name
+Specify the stream name PulseAudio will use when showing active streams,
+by default it is "record".
+
+@item sample_rate
+Specify the samplerate in Hz, by default 48kHz is used.
+
+@item channels
+Specify the channels in use, by default 2 (stereo) is set.
+
+@item frame_size
+Specify the number of bytes per frame, by default it is set to 1024.
+
+@item fragment_size
+Specify the minimal buffering fragment in PulseAudio, it will affect the
+audio latency. By default it is unset.
+@end table
+
+@subsection Examples
+Record a stream from default device:
@example
 ffmpeg -f pulse -i default /tmp/pulse.wav
@end example

-@subsection @var{server} AVOption
-
-The syntax is:
-@example
-server @var{server name}
-@end example
-
-Connects to a specific server.
-
-@subsection @var{name} AVOption
-
-The syntax is:
-@example
-name @var{application name}
-@end example
-
-Specify the application name pulse will use when showing active clients,
-by default it is the LIBAVFORMAT_IDENT string
-
-@subsection @var{stream_name} AVOption
-
-The syntax is:
-@example
-stream_name @var{stream name}
-@end example
-
-Specify the stream name pulse will use when showing active streams,
-by default it is "record"
-
-@subsection @var{sample_rate} AVOption
-
-The syntax is:
-@example
-sample_rate @var{samplerate}
-@end example
-
-Specify the samplerate in Hz, by default 48kHz is used.
-
-@subsection @var{channels} AVOption
-
-The syntax is:
-@example
-channels @var{N}
-@end example
-
-Specify the channels in use, by default 2 (stereo) is set.
-
-@subsection @var{frame_size} AVOption
-
-The syntax is:
-@example
-frame_size @var{bytes}
-@end example
-
-Specify the number of byte per frame, by default it is set to 1024.
-
-@subsection @var{fragment_size} AVOption
-
-The syntax is:
-@example
-fragment_size @var{bytes}
-@end example
-
-Specify the minimal buffering fragment in pulseaudio, it will affect the
-audio latency. By default it is unset.
-
@section sndio

 sndio input device.
@@ -590,8 +555,8 @@ 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), the device will always rely
-on libv4l2.
+@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
@@ -600,7 +565,7 @@ 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 devices, like TV cards, support one or more standards. It is possible
 to list all the supported standards using @command{-list_standards all}.
@@ -623,7 +588,7 @@ ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

@item
 Grab and record the input of a video4linux2 device, leave the
-framerate and size as previously set:
+frame rate and size as previously set:
@example
 ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
@end example
@@ -640,7 +605,8 @@ list of the supported standards, use the @option{list_standards}
 option.

@item channel
-Set the input channel number. Default to 0.
+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
@@ -651,11 +617,11 @@ 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
+This option allows one to select the input format, when several are
 available.

@item framerate
-Set the preferred video framerate.
+Set the preferred video frame rate.

@item list_formats
 List available formats (supported pixel formats, codecs, and frame
@@ -712,7 +678,7 @@ other filename will be interpreted as device number 0.

 X11 video input device.

-This device allows to capture a region of an X11 display.
+This device allows one to capture a region of an X11 display.

 The filename passed as input has the syntax:
@example
@@ -735,12 +701,12 @@ properties of your X11 display (e.g. grep for "name" or "dimensions").

 For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
-ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

 Grab at position @code{10,20}:
@example
-ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@end example

@subsection Options
@@ -761,17 +727,17 @@ zero) to the edge of region.

 For example:
@example
-ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

 To follow only when the mouse pointer reaches within 100 pixels to edge:
@example
-ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

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

@item show_region
 Show grabbed region on screen.
@@ -782,12 +748,12 @@ know what is being grabbed if only a portion of the screen is grabbed.

 For example:
@example
-ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@end example

 With @var{follow_mouse}:
@example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

@item video_size
@@ -1,4 +1,4 @@
-FFmpeg's bug/patch/feature request tracker manual
+FFmpeg's bug/feature request tracker manual
 =================================================

 NOTE: This is a draft.
@@ -11,7 +11,7 @@ existing issues can be done through a web interface.

 Issues can be different kinds of things we want to keep track of
 but that do not belong into the source tree itself. This includes
-bug reports, patches, feature requests and license violations. We
+bug reports, feature requests and license violations. We
 might add more items to this list in the future, so feel free to
 propose a new `type of issue' on the ffmpeg-devel mailing list if
 you feel it is worth tracking.
@@ -24,10 +24,13 @@ 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:
 -----
+art
+    Artwork such as photos, music, banners, and logos.
+
 bug / defect
    An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
    prevents it from behaving as intended.
@@ -41,20 +44,18 @@ feature request / enhancement
 license violation
    ticket to keep track of (L)GPL violations of ffmpeg by others

-patch
-    A patch as generated by diff which conforms to the patch submission and
-    development policy.
-
+sponsoring request
+    Developer requests for hardware, software, specifications, money,
+    refunds, etc.

 Priority:
 ---------
 critical
-    Bugs and patches which deal with data loss and security issues.
+    Bugs about data loss and security issues.
    No feature request can be critical.

 important
-    Bugs which make FFmpeg unusable for a significant number of users, and
-    patches fixing them.
+    Bugs which make FFmpeg unusable for a significant number of users.
    Examples here might be completely broken MPEG-4 decoding or a build issue
    on Linux.
    While broken 4xm decoding or a broken OS/2 build would not be important,
@@ -68,7 +69,7 @@ normal


 minor
-    Bugs and patches about things like spelling errors, "mp2" instead of
+    Bugs about things like spelling errors, "mp2" instead of
    "mp3" being shown and such.
    Feature requests about things few people want or which do not make a big
    difference.
@@ -103,13 +104,13 @@ This state implicates that the bug either has been reproduced or that
 reproduction is not needed as the bug is already understood.


-Type/Status/Substatus:
+Type/Status:
 ----------
-*/new/new
-    Initial state of new bugs, patches and feature requests submitted by
+*/new
+    Initial state of new bugs and feature requests submitted by
    users.

-*/open/open
+*/open
    Issues which have been briefly looked at and which did not look outright
    invalid.
    This implicates that no real more detailed state applies yet. Conversely,
@@ -117,9 +118,7 @@ Type/Status/Substatus:
    looked at.

 */closed/duplicate
-    Bugs, patches or feature requests which are duplicates.
-    Note that patches dealing with the same thing in a different way are not
-    duplicates.
+    Bugs or feature requests which are duplicates.
    Note, if you mark something as duplicate, do not forget setting the
    superseder so bug reports are properly linked.

@@ -134,7 +133,7 @@ Type/Status/Substatus:
 bug/closed/fixed
    Bugs which have to the best of our knowledge been fixed.

-bug/closed/wont_fix
+bug/closed/wontfix
    Bugs which we will not fix. Possible reasons include legality, high
    complexity for the sake of supporting obscure corner cases, speed loss
    for similarly esoteric purposes, et cetera.
@@ -148,33 +147,15 @@ bug/closed/works_for_me
    reproduction failed - that is the code seems to work correctly to the
    best of our knowledge.

-patch/open/approved
-    Patches which have been reviewed and approved by a developer.
-    Such patches can be applied anytime by any other developer after some
-    reasonable testing (compile + regression tests + does the patch do
-    what the author claimed).
-
-patch/open/needs_changes
-    Patches which have been reviewed and need changes to be accepted.
-
-patch/closed/applied
-    Patches which have been applied.
-
-patch/closed/rejected
-    Patches which have been rejected.
-
-feature_request/closed/implemented
+feature_request/closed/fixed
    Feature requests which have been implemented.

-feature_request/closed/wont_implement
+feature_request/closed/wontfix
    Feature requests which will not be implemented. The reasons here could
    be legal, philosophical or others.

-Note, please do not use type-status-substatus combinations other than the
-above without asking on ffmpeg-dev first!
-
 Note2, if you provide the requested info do not forget to remove the
-needs_more_info substatus.
+needs_more_info resolution.

 Component:
 ----------
@@ -16,7 +16,25 @@ 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).
+enumerations for pixel and sample formats). It is not a library for
+code needed by both libavcodec and libavformat.
+
+The goals for this library is to be:
+
+@table @strong
+@item Modular
+It should have few interdependencies and the possibility of disabling individual
+parts during @command{./configure}.
+
+@item Small
+Both sources and objects should be small.
+
+@item Efficient
+It should have low CPU and memory usage.
+
+@item Useful
+It should avoid useless features that almost no one needs.
+@end table

@c man end DESCRIPTION

@@ -20,7 +20,7 @@ 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
+example from a 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.

@@ -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
@@ -47,7 +47,14 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
      float_dsp_mips.c
      libm_mips.h
+* libavcodec/
+      fft_fixed_32.c
+      fft_init_table.c
+      fft_table.h
+      mdct_fixed_32.c
 * libavcodec/mips/
+      aaccoder_mips.c
+      aacpsy_mips.h
      ac3dsp_mips.c
      acelp_filters_mips.c
      acelp_vectors_mips.c
@@ -63,5 +70,6 @@ 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
@@ -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
@@ -18,6 +18,25 @@ enabled muxers.

 A description of some of the currently available muxers follows.

+@anchor{aiff}
+@section aiff
+
+Audio Interchange File Format muxer.
+
+@subsection Options
+
+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

@@ -32,6 +51,10 @@ The output of the muxer consists of a single line of the form:
 CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
 8 digits containing the CRC for all the decoded input frames.

+See also the @ref{framecrc} muxer.
+
+@subsection Examples
+
 For example to compute the CRC of the input, and store it in the file
@file{out.crc}:
@example
@@ -51,8 +74,6 @@ and the input video converted to MPEG-2 video, use the command:
 ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
@end example

-See also the @ref{framecrc} muxer.
-
@anchor{framecrc}
@section framecrc

@@ -72,6 +93,8 @@ packet of the form:
@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
 CRC of the packet.

+@subsection Examples
+
 For example to compute the CRC of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.crc}:
@@ -115,6 +138,8 @@ packet of the form:
@var{MD5} is a hexadecimal number representing the computed MD5 hash
 for the packet.

+@subsection Examples
+
 For example to compute the MD5 of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.md5}:
@@ -129,30 +154,89 @@ ffmpeg -i INPUT -f framemd5 -

 See also the @ref{md5} muxer.

+@anchor{gif}
+@section gif
+
+Animated GIF muxer.
+
+It accepts the following options:
+
+@table @option
+@item loop
+Set the number of times to loop the output. Use @code{-1} for no loop, @code{0}
+for looping indefinitely (default).
+
+@item final_delay
+Force the delay (expressed in centiseconds) after the last frame. Each frame
+ends with a delay until the next frame. The default is @code{-1}, which is a
+special value to tell the muxer to re-use the previous delay. In case of a
+loop, you might want to customize this value to mark a pause for instance.
+@end table
+
+For example, to encode a gif looping 10 times, with a 5 seconds delay between
+the loops:
+@example
+ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif
+@end example
+
+Note 1: if you wish to extract the frames in separate GIF files, you need to
+force the @ref{image2} muxer:
+@example
+ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
+@end example
+
+Note 2: the GIF format has a very small time base: the delay between two frames
+can not be smaller than one centi second.
+
@anchor{hls}
@section hls

 Apple HTTP Live Streaming muxer that segments MPEG-TS according to
-the HTTP Live Streaming specification.
+the HTTP Live Streaming (HLS) 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.

+For example, to convert an input file with @command{ffmpeg}:
@example
 ffmpeg -i in.nut out.m3u8
@end example

+See also the @ref{segment} muxer, which provides a more generic and
+flexible implementation of a segmenter, and can be used to perform HLS
+segmentation.
+
+@subsection Options
+
+This muxer supports the following options:
+
@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}.
+@item hls_time @var{seconds}
+Set the segment length in seconds. Default value is 2.
+
+@item hls_list_size @var{size}
+Set the maximum number of playlist entries. If set to 0 the list file
+will contain all the segments. Default value is 5.
+
+@item hls_wrap @var{wrap}
+Set the number after which the segment filename number (the number
+specified in each segment file) wraps. If set to 0 the number will be
+never wrapped. Default value is 0.
+
+This option is useful to avoid to fill the disk with many segment
+files, and limits the maximum number of segment files written to disk
+to @var{wrap}.
+
+@item start_number @var{number}
+Start the playlist sequence number from @var{number}. Default value is
+0.
+
+Note that the playlist sequence number must be unique for each segment
+and it is not to be confused with the segment filename sequence number
+which can be cyclic, for example if the @option{wrap} option is
+specified.
@end table

@anchor{ico}
@@ -218,6 +302,8 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
 form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
 etc.

+@subsection Examples
+
 The following example shows how to use @command{ffmpeg} for creating a
 sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
 taking one image every second from the input video:
@@ -240,14 +326,32 @@ 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.
+The @option{strftime} option allows you to expand the filename with
+date and time information. Check the documentation of
+the @code{strftime()} function for the syntax.

-@item updatefirst 1|0
-If set to 1, update the first written image file again and
-again. Default value is 0.
+For example to generate image files from the @code{strftime()}
+"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command
+can be used:
+@example
+ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
+@end example
+
+@subsection Options
+
+@table @option
+@item start_number
+Start the sequence from the specified number. Default value is 1. Must
+be a non-negative number.
+
+@item update
+If set to 1, the filename will always be interpreted as just a
+filename, not a pattern, and the corresponding file will be continuously
+overwritten with new images. Default value is 0.
+
+@item strftime
+If set to 1, expand the filename with date and time information from
+@code{strftime()}. Default value is 0.
@end table

 The image muxer supports the .Y.U.V image file format. This format is
@@ -256,6 +360,92 @@ 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.
+
+@subsection Metadata
+
+The recognized metadata settings in this muxer are:
+
+@table @option
+@item title
+Set title name provided to a single track.
+
+@item language
+Specify the language of the track in the Matroska languages form.
+
+The language can be either the 3 letters bibliographic ISO-639-2 (ISO
+639-2/B) form (like "fre" for French), or a language code mixed with a
+country code for specialities in languages (like "fre-ca" for Canadian
+French).
+
+@item stereo_mode
+Set stereo 3D video layout of two views in a single video track.
+
+The following values are recognized:
+@table @samp
+@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
+
+@subsection Options
+
+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

@@ -283,7 +473,9 @@ ffmpeg -i INPUT -f md5 -

 See also the @ref{framemd5} muxer.

-@section MOV/MP4/ISMV
+@section mov, mp4, ismv
+
+MOV/MP4/ISMV (Smooth Streaming) muxer.

 The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
 file has all the metadata about all packets stored in one location
@@ -299,6 +491,8 @@ very long files (since writing normal MOV/MP4 files stores info about
 every single packet in memory until the file is closed). The downside
 is that it is less compatible with other applications.

+@subsection Options
+
 Fragmentation is enabled by setting one of the AVOptions that define
 how to cut the file into fragments:

@@ -348,25 +542,78 @@ 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
+Run a second pass moving the index (moov atom) to the beginning 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

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

+@section 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. Setting
+@code{id3v2_version} to 0 will disable the ID3v2 header completely. The legacy
+ID3v1 tag is not written by default, but may be enabled with the
+@code{write_id3v1} option.
+
+The muxer may also write 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 Xing frame is written if the output stream is seekable and if the
+@code{write_xing} option is set to 1 (the default).
+
+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
+
+Write a "clean" MP3 without any extra features:
+@example
+ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3
+@end example
+
@section mpegts

 MPEG transport stream muxer.

 This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

+The recognized metadata settings in mpegts muxer are @code{service_provider}
+and @code{service_name}. If they are not set the default for
+@code{service_provider} is "FFmpeg" and the default for
+@code{service_name} is "Service01".
+
+@subsection Options
+
 The muxer options are:

@table @option
@@ -383,12 +630,43 @@ Set the service_id (default 0x0001) also known as program in DVB.
 Set the first PID for PMT (default 0x1000, max 0x1f00).
@item -mpegts_start_pid @var{number}
 Set the first PID for data packets (default 0x0100, max 0x0f00).
+@item -mpegts_m2ts_mode @var{number}
+Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
+@item -muxrate @var{number}
+Set muxrate.
+@item -pes_payload_size @var{number}
+Set minimum PES packet payload in bytes.
+@item -mpegts_flags @var{flags}
+Set flags (see below).
+@item -mpegts_copyts @var{number}
+Preserve original timestamps, if value is set to 1. Default value is -1, which
+results in shifting timestamps so that they start from 0.
+@item -tables_version @var{number}
+Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
+This option allows updating stream structure so that standard consumer may
+detect the change. To do so, reopen output AVFormatContext (in case of API
+usage) or restart ffmpeg instance, cyclically changing tables_version value:
+@example
+ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
+ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
+...
+ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
+ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
+ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
+...
+@end example
@end table

-The recognized metadata settings in mpegts muxer are @code{service_provider}
-and @code{service_name}. If they are not set the default for
-@code{service_provider} is "FFmpeg" and the default for
-@code{service_name} is "Service01".
+Option mpegts_flags may take a set of such flags:
+
+@table @option
+@item resend_headers
+Reemit PAT/PMT before writing the next packet.
+@item latm
+Use LATM packetization for AAC.
+@end table
+
+@subsection Example

@example
 ffmpeg -i file.mpg -c copy \
@@ -424,74 +702,27 @@ 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
-
+@anchor{segment}
@section segment, stream_segment, ssegment

 Basic stream segmenter.

-The segmenter muxer outputs streams to a number of separate files of nearly
+This muxer outputs streams to a number of separate files of nearly
 fixed duration. Output filename pattern can be set in a fashion similar to
@ref{image2}.

@@ -513,17 +744,24 @@ The segment muxer works best with a single constant frame rate video.

 Optionally it can generate a list of the created segments, by setting
 the option @var{segment_list}. The list type is specified by the
-@var{segment_list_type} option.
+@var{segment_list_type} option. The entry filenames in the segment
+list are set by default to the basename of the corresponding segment
+files.
+
+See also the @ref{hls} muxer, which provides a more specific
+implementation for HLS segmentation.
+
+@subsection Options

 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
+If @var{specifier} is set to @code{auto}, the reference is chosen
 automatically. Otherwise it must be a stream specifier (see the ``Stream
 specifiers'' chapter in the ffmpeg manual) which specifies the
-reference stream. The default value is ``auto''.
+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
@@ -537,7 +775,7 @@ listfile is generated.
 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).

@@ -545,18 +783,20 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
@end table

-Default value is @code{cache}.
-
@item segment_list_size @var{size}
 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}
+@item segment_list_entry_prefix @var{prefix}
+Set @var{prefix} to prepend to the name of each entry filename. By
+default no prefix is applied.
+
+@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.

@@ -577,7 +817,7 @@ 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
@@ -623,7 +863,7 @@ In particular may be used in combination with the @file{ffmpeg} option
@var{force_key_frames} may not be set accurately because of rounding
 issues, with the consequence that a key frame time may result set just
 before the specified time. For constant frame rate videos a value of
-1/2*@var{frame_rate} should address the worst case mismatch between
+1/(2*@var{frame_rate}) should address the worst case mismatch between
 the specified time and the time set by @var{force_key_frames}.

@item segment_times @var{times}
@@ -650,6 +890,10 @@ 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.
+
+@item initial_offset @var{offset}
+Specify timestamp offset to apply to the output packet timestamps. The
+argument must be a time duration specification, and defaults to 0.
@end table

@subsection Examples
@@ -671,9 +915,9 @@ 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 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
@@ -684,7 +928,7 @@ required.

@item
 Segment the input file by splitting the input file according to the
-frame numbers sequence specified with the @var{segment_frames} option:
+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
@@ -705,57 +949,6 @@ ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
@end example
@end itemize

-@section mp3
-
-The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
-optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
-not written by default, but may be enabled with the @code{write_id3v1} option.
-
-For seekable output the muxer also writes a Xing frame at the beginning, which
-contains the number of frames in the file. It is useful for computing duration
-of VBR files.
-
-The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
-are supplied to the muxer in form of a video stream with a single packet. There
-can be any number of those streams, each will correspond to a single APIC frame.
-The stream metadata tags @var{title} and @var{comment} map to APIC
-@var{description} and @var{picture type} respectively. See
-@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
-
-Note that the APIC frames must be written at the beginning, so the muxer will
-buffer the audio frames until it gets all the pictures. It is therefore advised
-to provide the pictures as soon as possible to avoid excessive buffering.
-
-Examples:
-
-Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
-@example
-ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
-@end example
-
-To attach a picture to an mp3 file select both the audio and the picture stream
-with @code{map}:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
-@end example
-
-@section ogg
-
-Ogg container muxer.
-
-@table @option
-@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
-
@section tee

 The tee muxer can be used to write the same data to several files or any
@@ -771,22 +964,71 @@ to feed the same packets to several muxers directly.
 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).
+escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
+section in the ffmpeg-utils(1) manual,ffmpeg-utils}).

-Options can be specified for each slave by prepending them as a list of
+Muxer 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.

-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):
+The following special options are also recognized:
+@table @option
+@item f
+Specify the format name. Useful if it cannot be guessed from the
+output name suffix.

+@item bsfs[/@var{spec}]
+Specify a list of bitstream filters to apply to the specified
+output.
+
+It is possible to specify to which streams a given bitstream filter
+applies, by appending a stream specifier to the option separated by
+@code{/}. @var{spec} must be a stream specifier (see @ref{Format
+stream specifiers}).  If the stream specifier is not specified, the
+bistream filters will be applied to all streams in the output.
+
+Several bitstream filters can be specified, separated by ",".
+
+@item select
+Select the streams that should be mapped to the slave output,
+specified by a stream specifier. If not specified, this defaults to
+all the input streams.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+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):
@example
 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

+@item
+Use @command{ffmpeg} to encode the input, and send the output
+to three different destinations. The @code{dump_extra} bitstream
+filter is used to add extradata information to all the output video
+keyframes packets, as requested by the MPEG-TS format. The select
+option is applied to @file{out.aac} in order to make it contain only
+audio packets.
+@example
+ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
+       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
+@end example
+
+@item
+As below, but select only stream @code{a:1} for the audio output. Note
+that a second level escaping must be performed, as ":" is a special
+character used to separate options.
+@example
+ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
+       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
+@end example
+@end itemize
+
 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.
@@ -1,7 +1,7 @@
@chapter Output Devices
@c man begin OUTPUT DEVICES

-Output devices are configured elements in FFmpeg which allow to write
+Output devices are configured elements in FFmpeg that can write
 multimedia data to an output device attached to your system.

 When you configure your FFmpeg build, all the supported output devices
@@ -22,11 +22,27 @@ A description of the currently available output devices follows.

 ALSA (Advanced Linux Sound Architecture) output device.

+@subsection Examples
+
+@itemize
+@item
+Play a file on default ALSA device:
+@example
+ffmpeg -i INPUT -f alsa default
+@end example
+
+@item
+Play a file on soundcard 1, audio device 7:
+@example
+ffmpeg -i INPUT -f alsa hw:1,7
+@end example
+@end itemize
+
@section caca

 CACA output device.

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

@@ -104,15 +120,189 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
@end example
@end itemize

+@section decklink
+
+The decklink output device provides playback capabilities for Blackmagic
+DeckLink devices.
+
+To enable this output device, you need the Blackmagic DeckLink SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+On Windows, you need to run the IDL files through @command{widl}.
+
+DeckLink is very picky about the formats it supports. Pixel format is always
+uyvy422, framerate and video size must be determined for your device with
+@command{-list_formats 1}. Audio sample rate is always 48 kHz.
+
+@subsection Options
+
+@table @option
+
+@item list_devices
+If set to @option{true}, print a list of devices and exit.
+Defaults to @option{false}.
+
+@item list_formats
+If set to @option{true}, print a list of supported formats and exit.
+Defaults to @option{false}.
+
+@item preroll
+Amount of time to preroll video in seconds.
+Defaults to @option{0.5}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+List output devices:
+@example
+ffmpeg -i test.avi -f decklink -list_devices 1 dummy
+@end example
+
+@item
+List supported formats:
+@example
+ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
+@end example
+
+@item
+Play video clip:
+@example
+ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
+@end example
+
+@item
+Play video clip with non-standard framerate or video size:
+@example
+ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
+@end example
+
+@end itemize
+
+@section fbdev
+
+Linux framebuffer output device.
+
+The Linux framebuffer is a graphic hardware-independent abstraction
+layer to show graphics on a computer monitor, typically on the
+console. It is accessed through a file device node, usually
+@file{/dev/fb0}.
+
+For more detailed information read the file
+@file{Documentation/fb/framebuffer.txt} included in the Linux source tree.
+
+@subsection Options
+@table @option
+
+@item xoffset
+@item yoffset
+Set x/y coordinate of top left corner. Default is 0.
+@end table
+
+@subsection Examples
+Play a file on framebuffer device @file{/dev/fb0}.
+Required pixel format depends on current framebuffer settings.
+@example
+ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
+@end example
+
+See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
+
+@section opengl
+OpenGL output device.
+
+To enable this output device you need to configure FFmpeg with @code{--enable-opengl}.
+
+This output device allows one to render to OpenGL context.
+Context may be provided by application or default SDL window is created.
+
+When device renders to external context, application must implement handlers for following messages:
+@code{AV_CTL_MESSAGE_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread.
+@code{AV_CTL_MESSAGE_PREPARE_WINDOW_BUFFER} - make OpenGL context current.
+@code{AV_CTL_MESSAGE_DISPLAY_WINDOW_BUFFER} - swap buffers.
+@code{AV_CTL_MESSAGE_DESTROY_WINDOW_BUFFER} - destroy OpenGL context.
+Application is also required to inform a device about current resolution by sending @code{AV_DEVICE_WINDOW_RESIZED} message.
+
+@subsection Options
+@table @option
+
+@item background
+Set background color. Black is a default.
+@item no_window
+Disables default SDL window when set to non-zero value.
+Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set.
+@item window_title
+Set the SDL window title, if not specified default to the filename specified for the output device.
+Ignored when @option{no_window} is set.
+
+@end table
+
+@subsection Examples
+Play a file on SDL window using OpenGL rendering:
+@example
+ffmpeg  -i INPUT -f opengl "window title"
+@end example
+
@section oss

 OSS (Open Sound System) output device.

+@section pulse
+
+PulseAudio output device.
+
+To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
+
+More information about PulseAudio can be found on @url{http://www.pulseaudio.org}
+
+@subsection Options
+@table @option
+
+@item server
+Connect to a specific PulseAudio server, specified by an IP address.
+Default server is used when not provided.
+
+@item name
+Specify the application name PulseAudio will use when showing active clients,
+by default it is the @code{LIBAVFORMAT_IDENT} string.
+
+@item stream_name
+Specify the stream name PulseAudio will use when showing active streams,
+by default it is set to the specified output name.
+
+@item device
+Specify the device to use. Default device is used when not provided.
+List of output devices can be obtained with command @command{pactl list sinks}.
+
+@item buffer_size
+@item buffer_duration
+Control the size and duration of the PulseAudio buffer. A small buffer
+gives more control, but requires more frequent updates.
+
+@option{buffer_size} specifies size in bytes while
+@option{buffer_duration} specifies duration in milliseconds.
+
+When both options are provided then the highest value is used
+(duration is recalculated to bytes using stream parameters). If they
+are set to 0 (which is default), the device will use the default
+PulseAudio duration value. By default PulseAudio set buffer duration
+to around 2 seconds.
+@end table
+
+@subsection Examples
+Play a file on default device on default server:
+@example
+ffmpeg  -i INPUT -f pulse "stream name"
+@end example
+
@section sdl

 SDL (Simple DirectMedia Layer) output device.

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

@@ -139,6 +329,20 @@ Set the SDL window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
 If not specified it defaults to the size of the input video,
 downscaled according to the aspect ratio.
+
+@item window_fullscreen
+Set fullscreen mode when non-zero value is provided.
+Default value is zero.
+@end table
+
+@subsection Interactive commands
+
+The window created by the device can be controlled through the
+following interactive commands.
+
+@table @key
+@item q, ESC
+Quit the device immediately.
@end table

@subsection Examples
@@ -153,4 +357,69 @@ 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 one 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
@@ -51,8 +51,9 @@ The toolchain provided with Xcode is sufficient to build the basic
 unacelerated code.

 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
+@url{https://github.com/FFmpeg/gas-preprocessor} or
@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
-assembler functions. Just download the Perl script and put it somewhere
+assembler functions. Put the Perl script somewhere
 in your PATH, FFmpeg's configure will pick it up automatically.

 Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
@@ -106,44 +107,53 @@ libavformat) as DLLs.

@end itemize

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

-FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and
-wrapper.
+FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
+and wrapper, or with MSVC 2013 and ICL natively.

 You will need the following prerequisites:

@itemize
-@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+(if using MSVC 2012 or earlier)
@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
+(if using MSVC 2012 or earlier)
@item @uref{http://www.mingw.org/, MSYS}
@item @uref{http://yasm.tortall.net/, YASM}
@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
 you want to run @uref{fate.html, FATE}.
@end itemize

-To set up a proper MSVC environment in MSYS, you simply need to run
-@code{msys.bat} from the Visual Studio command prompt.
+To set up a proper environment in MSYS, you need to run @code{msys.bat} from
+the Visual Studio or Intel Compiler command prompt.

-Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
-somewhere in your @code{PATH}.
+Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
+earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
+@code{PATH} as well.

-Next, make sure @code{inttypes.h} and any other headers and libs you want to use
-are located in a spot that MSVC can see. Do so by modifying the @code{LIB} and
-@code{INCLUDE} environment variables to include the @strong{Windows} paths to
-these directories. Alternatively, you can try and use the
-@code{--extra-cflags}/@code{--extra-ldflags} configure options.
+Next, make sure any other headers and libs you want to use, such as zlib, 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-style}
+paths to these directories. Alternatively, you can try and use the
+@code{--extra-cflags}/@code{--extra-ldflags} configure options. If using MSVC
+2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.

 Finally, run:

@example
+For MSVC:
 ./configure --toolchain=msvc
+
+For ICL:
+./configure --toolchain=icl
+
 make
 make install
@end example

 If you wish to compile shared libraries, add @code{--enable-shared} to your
-configure options. Note that due to the way MSVC handles DLL imports and
+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.

@@ -173,7 +183,14 @@ erroneously included when building FFmpeg.
 can see.
@end enumerate

-@item FFmpeg has been tested with Visual Studio 2010 and 2012, Pro and Express.
+@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 Visual Studio 2013 Pro and Express
+@item Intel Composer XE 2013
+@item Intel Composer XE 2013 SP1
+@end itemize
 Anything else is not officially supported.

@end itemize
@@ -184,16 +201,7 @@ If you plan to link with MSVC-built static libraries, you will need
 to make sure you have @code{Runtime Library} set to
@code{Multi-threaded (/MT)} in your project's settings.

-FFmpeg headers do not declare global data for Windows DLLs through the usual
-dllexport/dllimport interface. Such data will be exported properly while
-building, but to use them in your MSVC code you will have to edit the
-appropriate headers and mark the data as dllimport. For example, in
-libavutil/pixdesc.h you should have:
-@example
-extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
-@end example
-
-You will also need to define @code{inline} to something MSVC understands:
+You will need to define @code{inline} to something MSVC understands:
@example
 #define inline __inline
@end example
@@ -1,20 +1,20 @@
 /*
 * Copyright (c) 2012 Anton Khirnov
 *
- * 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
 */

@@ -27,7 +27,9 @@
 #include <float.h>

 #include "libavformat/avformat.h"
+#include "libavformat/options_table.h"
 #include "libavcodec/avcodec.h"
+#include "libavcodec/options_table.h"
 #include "libavutil/opt.h"

 static void print_usage(void)
@@ -39,6 +41,9 @@ 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;
@@ -93,18 +98,14 @@ static void show_opts(const AVOption *opts, int per_stream)

 static void show_format_opts(void)
 {
-#include "libavformat/options_table.h"
-
    printf("@section Format AVOptions\n");
-    show_opts(options, 0);
+    show_opts(avformat_options, 0);
 }

 static void show_codec_opts(void)
 {
-#include "libavcodec/options_table.h"
-
    printf("@section Codec AVOptions\n");
-    show_opts(options, 1);
+    show_opts(avcodec_options, 1);
 }

 int main(int argc, char **argv)
@@ -1,8 +1,8 @@
@chapter Protocols
@c man begin PROTOCOLS

-Protocols are configured elements in FFmpeg which allow to access
-resources which require the use of a particular protocol.
+Protocols are configured elements in FFmpeg that enable access to
+resources that require specific protocols.

 When you configure your FFmpeg build, all the supported protocols are
 enabled by default. You can list all available ones using the
@@ -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 +85,25 @@ 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}.
@@ -88,7 +117,19 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////

 File access protocol.

-Allow to read from or read to a file.
+Allow to read from or write to a file.
+
+A file URL can have the form:
+@example
+file:@var{filename}
+@end example
+
+where @var{filename} is the path of the file to read.
+
+An URL that does not have a protocol prefix will be assumed to be a
+file URL. Depending on the build, an URL that looks like a Windows
+path with the drive letter at the beginning will also be assumed to be
+a file URL (usually not the case in builds for unix-like systems).

 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
 use the command:
@@ -96,9 +137,53 @@ use the command:
 ffmpeg -i file:input.mpeg output.mpeg
@end example

-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".
+This protocol accepts the following options:
+
+@table @option
+@item truncate
+Truncate existing files on write, if set to 1. A value of 0 prevents
+truncating. Default value is 1.
+
+@item blocksize
+Set I/O operation maximum block size, in bytes. Default value is
+@code{INT_MAX}, which results in not limiting the requested block size.
+Setting this value reasonably low improves user termination request reaction
+time, which is valuable for files on slow medium.
+@end table
+
+@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

@@ -128,7 +213,7 @@ m3u8 files.

 HTTP (Hyper Text Transfer Protocol).

-This protocol accepts the following options.
+This protocol accepts the following options:

@table @option
@item seekable
@@ -138,37 +223,60 @@ 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.
+If set to 1 use chunked Transfer-Encoding for posts, default is 1.
+
+@item content_type
+Set a specific content type for the POST messages.

@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.
+Use persistent connections if set to 1, default is 0.

@item post_data
 Set custom HTTP post data.

+@item user-agent
+@item user_agent
+Override the User-Agent header. If not specified the protocol will use a
+string describing the libavformat build. ("Lavf/<version>")
+
@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.
+Export the 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 by 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. It should be
+polled in regular intervals by applications interested in mid-stream metadata
+updates.

@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.
+
+@item offset
+Set initial byte offset.
+
+@item end_offset
+Try to limit the request to bytes preceding this offset.
@end table

@subsection HTTP Cookies
@@ -248,6 +356,16 @@ ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
@end example

+This protocol accepts the following options:
+
+@table @option
+@item blocksize
+Set I/O operation maximum block size, in bytes. Default value is
+@code{INT_MAX}, which results in not limiting the requested block size.
+Setting this value reasonably low improves user termination request reaction
+time, which is valuable if data transmission is slow.
+@end table
+
 Note that some formats (typically MOV), require the output protocol to
 be seekable, so they will fail with the pipe output protocol.

@@ -260,12 +378,18 @@ content across a TCP/IP network.

 The required syntax is:
@example
-rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
+rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
@end example

 The accepted parameters are:
@table @option

+@item username
+An optional username (mostly for publishing).
+
+@item password
+An optional password (mostly for publishing).
+
@item server
 The address of the RTMP server.

@@ -316,7 +440,8 @@ times to construct arbitrary AMF sequences.

@item rtmp_flashver
 Version of the Flash plugin used to run the SWF player. The default
-is LNX 9,0,124,2.
+is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
+<libavformat version>).)

@item rtmp_flush_interval
 Number of packets flushed in the same request (RTMPT only). The default
@@ -366,6 +491,12 @@ For example to read with @command{ffplay} a multimedia resource named
 ffplay rtmp://myserver/vod/sample
@end example

+To publish to a password protected server, passing the playpath and
+app names separately:
+@example
+ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
+@end example
+
@section rtmpe

 Encrypted Real-Time Messaging Protocol.
@@ -406,7 +537,43 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
 for streaming multimedia content within HTTPS requests to traverse
 firewalls.

-@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
+@section libssh
+
+Secure File Transfer Protocol via libssh
+
+Allow to read from or write to remote resources using SFTP protocol.
+
+Following syntax is required.
+
+@example
+sftp://[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 truncate
+Truncate existing files on write, if set to 1. A value of 0 prevents
+truncating. Default value is 1.
+
+@item private_key
+Specify the path of the file containing private key to use during authorization.
+By default libssh searches for keys in the @file{~/.ssh/} directory.
+
+@end table
+
+Example: Play a file stored on remote server.
+
+@example
+ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
+@end example
+
+@section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte

 Real-Time Messaging Protocol and its variants supported through
 librtmp.
@@ -448,10 +615,75 @@ ffplay "rtmp://myserver/live/mystream live=1"

@section rtp

-Real-Time Protocol.
+Real-time Transport Protocol.
+
+The required syntax for an RTP URL is:
+rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
+
+@var{port} specifies the RTP port to use.
+
+The following URL options are supported:
+
+@table @option
+
+@item ttl=@var{n}
+Set the TTL (Time-To-Live) value (for multicast only).
+
+@item rtcpport=@var{n}
+Set the remote RTCP port to @var{n}.
+
+@item localrtpport=@var{n}
+Set the local RTP port to @var{n}.
+
+@item localrtcpport=@var{n}'
+Set the local RTCP port to @var{n}.
+
+@item pkt_size=@var{n}
+Set max packet size (in bytes) to @var{n}.
+
+@item connect=0|1
+Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
+to 0).
+
+@item sources=@var{ip}[,@var{ip}]
+List allowed source IP addresses.
+
+@item block=@var{ip}[,@var{ip}]
+List disallowed (blocked) source IP addresses.
+
+@item write_to_source=0|1
+Send packets to the source address of the latest received packet (if
+set to 1) or to a default remote address (if set to 0).
+
+@item localport=@var{n}
+Set the local RTP port to @var{n}.
+
+This is a deprecated option. Instead, @option{localrtpport} should be
+used.
+
+@end table
+
+Important notes:
+
+@enumerate
+
+@item
+If @option{rtcpport} is not set the RTCP port will be set to the RTP
+port value plus 1.
+
+@item
+If @option{localrtpport} (the local RTP port) is not set any available
+port will be used for the local RTP and RTCP ports.
+
+@item
+If @option{localrtcpport} (the local RTCP port) is not set it will be
+set to the the local RTP port value plus 1.
+@end enumerate

@section rtsp

+Real-Time Streaming Protocol.
+
 RTSP is not technically a protocol handler in libavformat, it is a demuxer
 and muxer. The demuxer supports both normal RTSP (with data transferred
 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
@@ -466,14 +698,22 @@ The required syntax for a RTSP url is:
 rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example

-The following options (set on the @command{ffmpeg}/@command{ffplay} command
-line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
-are supported:
+Options can be set on the @command{ffmpeg}/@command{ffplay} command
+line, or set in code via @code{AVOption}s or in
+@code{avformat_open_input}.

-Flags for @code{rtsp_transport}:
+The following options are supported.

@table @option
+@item initial_pause
+Do not start playing the stream immediately if set to 1. Default value
+is 0.

+@item rtsp_transport
+Set RTSP trasport protocols.
+
+It accepts the following values:
+@table @samp
@item udp
 Use UDP as lower transport protocol.

@@ -491,17 +731,56 @@ passing proxies.

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

-Flags for @code{rtsp_flags}:
+@item rtsp_flags
+Set RTSP flags.

-@table @option
+The following values are accepted:
+@table @samp
@item filter_src
 Accept packets only from negotiated peer address and port.
@item listen
 Act as a server, listening for an incoming connection.
@end table

+Default value is @samp{none}.
+
+@item allowed_media_types
+Set media types to accept from the server.
+
+The following flags are accepted:
+@table @samp
+@item video
+@item audio
+@item data
+@end table
+
+By default it accepts all media types.
+
+@item min_port
+Set minimum local UDP port. Default value is 5000.
+
+@item max_port
+Set maximum local UDP port. Default value is 65000.
+
+@item timeout
+Set maximum timeout (in seconds) to wait for incoming connections.
+
+A value of -1 mean infinite (default). This option implies the
+@option{rtsp_flags} set to @samp{listen}.
+
+@item reorder_queue_size
+Set number of packets to buffer for handling of reordered packets.
+
+@item stimeout
+Set socket TCP I/O timeout in micro seconds.
+
+@item user-agent
+Override User-Agent header. If not specified, it default to the
+libavformat identifier string.
+@end table
+
 When receiving data over UDP, the demuxer tries to reorder received packets
 (since they may arrive out of order, or packets may get lost totally). This
 can be disabled by setting the maximum demuxing delay to zero (via
@@ -512,31 +791,36 @@ streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.

-Example command lines:
+@subsection Examples

-To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
+The following examples all make use of the @command{ffplay} and
+@command{ffmpeg} tools.

+@itemize
+@item
+Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@example
 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example

-To watch a stream tunneled over HTTP:
-
+@item
+Watch a stream tunneled over HTTP:
@example
 ffplay -rtsp_transport http rtsp://server/video.mp4
@end example

-To send a stream in realtime to a RTSP server, for others to watch:
-
+@item
+Send a stream in realtime to a RTSP server, for others to watch:
@example
 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example

-To receive a stream in realtime:
-
+@item
+Receive a stream in realtime:
@example
 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@end example
+@end itemize

@section sap

@@ -630,57 +914,129 @@ 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.
+Transmission Control Protocol.

 The required syntax for a TCP url is:
@example
 tcp://@var{hostname}:@var{port}[?@var{options}]
@end example

-@table @option
+@var{options} contains a list of &-separated options of the form
+@var{key}=@var{val}.

-@item listen
-Listen for an incoming connection
+The list of supported options follows.
+
+@table @option
+@item listen=@var{1|0}
+Listen for an incoming connection. Default value is 0.

@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
-In write mode: if socket cannot be written in more than this time interval, raise error.
-This also sets timeout on TCP connection establishing.
+Set raise error timeout, expressed in microseconds.

+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
+
+@item listen_timeout=@var{microseconds}
+Set listen timeout, expressed in microseconds.
+@end table
+
+The following example shows how to setup a listening TCP connection
+with @command{ffmpeg}, which is then accessed with @command{ffplay}:
@example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
@end example

-@end table
-
@section tls

-Transport Layer Security/Secure Sockets Layer
+Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

 The required syntax for a TLS/SSL url is:
@example
 tls://@var{hostname}:@var{port}[?@var{options}]
@end example

+The following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+
@table @option

-@item listen
-Act as a server, listening for an incoming connection.
+@item ca_file, cafile=@var{filename}
+A file containing certificate authority (CA) root certificates to treat
+as trusted. If the linked TLS library contains a default this might not
+need to be specified for verification to work, but not all libraries and
+setups have defaults built in.
+The file must be in OpenSSL PEM format.

-@item cafile=@var{filename}
-Certificate authority file. The file must be in OpenSSL PEM format.
+@item tls_verify=@var{1|0}
+If enabled, try to verify the peer that we are communicating with.
+Note, if using OpenSSL, this currently only makes sure that the
+peer certificate is signed by one of the root certificates in the CA
+database, but it does not validate that the certificate actually
+matches the host name we are trying to connect to. (With GnuTLS,
+the host name is validated as well.)

-@item cert=@var{filename}
-Certificate file. The file must be in OpenSSL PEM format.
+This is disabled by default since it requires a CA database to be
+provided by the caller in many cases.

-@item key=@var{filename}
-Private key file.
+@item cert_file, cert=@var{filename}
+A file containing a certificate to use in the handshake with the peer.
+(When operating as server, in listen mode, this is more often required
+by the peer, while client certificates only are mandated in certain
+setups.)

-@item verify=@var{0|1}
-Verify the peer's certificate.
+@item key_file, key=@var{filename}
+A file containing the private key for the certificate.
+
+@item listen=@var{1|0}
+If enabled, listen for connections on the provided port, and assume
+the server role in the handshake instead of the client role.

@end table

@@ -702,7 +1058,7 @@ ffplay tls://@var{hostname}:@var{port}

 User Datagram Protocol.

-The required syntax for a UDP url is:
+The required syntax for an UDP URL is:
@example
 udp://@var{hostname}:@var{port}[?@var{options}]
@end example
@@ -710,14 +1066,13 @@ udp://@var{hostname}:@var{port}[?@var{options}]
@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.

 In case threading is enabled on the system, a circular buffer is used
-to store the incoming data, which allows to reduce loss of data due to
+to store the incoming data, which allows one to reduce loss of data due to
 UDP socket buffer overruns. The @var{fifo_size} and
@var{overrun_nonfatal} options are related to this buffer.

 The list of supported options follows.

@table @option
-
@item buffer_size=@var{size}
 Set the UDP socket buffer size in bytes. This is used both for the
 receiving and the sending buffer size.
@@ -767,24 +1122,53 @@ Survive in case of UDP receiving circular buffer overrun. Default
 value is 0.

@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
+Set raise error timeout, expressed in microseconds.
+
+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
@end table

-Some usage examples of the UDP protocol with @command{ffmpeg} follow.
+@subsection Examples

-To stream over UDP to a remote endpoint:
+@itemize
+@item
+Use @command{ffmpeg} to stream over UDP to a remote endpoint:
@example
 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example

-To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
+@item
+Use @command{ffmpeg} to stream in mpegts format over UDP using 188
+sized UDP packets, using a large input buffer:
@example
 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
@end example

-To receive over UDP from a remote endpoint:
+@item
+Use @command{ffmpeg} to receive over UDP from a remote endpoint:
@example
-ffmpeg -i udp://[@var{multicast-address}]:@var{port}
+ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
+@end example
+@end itemize
+
+@section unix
+
+Unix local socket
+
+The required syntax for a Unix socket URL is:
+
+@example
+unix://@var{filepath}
@end example

+The following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+
+@table @option
+@item timeout
+Timeout in ms.
+@item listen
+Create the Unix socket in listening mode.
+@end table
+
@c man end PROTOCOLS
@@ -0,0 +1,232 @@
+@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
+@item ocl, out_channel_layout
+Set the input/output channel layout.
+
+See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
+for the required syntax.
+
+@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 rematrix_maxval
+Set maximum output value for rematrixing.
+This can be used to prevent clipping vs. preventing volumn reduction
+A value of 1.0 prevents cliping.
+
+@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
@@ -0,0 +1,119 @@
+@anchor{scaler_options}
+@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
+
+@anchor{sws_flags}
+@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 bicublin
+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.
+
+@item sws_dither
+Set the dithering algorithm. Accepts one of the following
+values. Default value is @samp{auto}.
+
+@table @samp
+@item auto
+automatic choice
+
+@item none
+no dithering
+
+@item bayer
+bayer dither
+
+@item ed
+error diffusion dither
+@end table
+
+@end table
+
+@c man end SCALER OPTIONS
@@ -50,8 +50,10 @@ header:
        temporal_decomposition_count    u   header_state
        spatial_decomposition_count     u   header_state
        colorspace_type                 u   header_state
-        chroma_h_shift                  u   header_state
-        chroma_v_shift                  u   header_state
+        if (nb_planes > 2) {
+            chroma_h_shift              u   header_state
+            chroma_v_shift              u   header_state
+        }
        spatial_scalability             b   header_state
        max_ref_frames-1                u   header_state
        qlogs
@@ -59,7 +61,7 @@ header:
    if(!keyframe){
        update_mc                       b   header_state
        if(update_mc){
-            for(plane=0; plane<2; plane++){
+            for(plane=0; plane<nb_plane_types; plane++){
                diag_mc                 b   header_state
                htaps/2-1               u   header_state
                for(i= p->htaps/2; i; i--)
@@ -80,7 +82,7 @@ header:
    block_max_depth                     s   header_state

 qlogs:
-    for(plane=0; plane<2; plane++){
+    for(plane=0; plane<nb_plane_types; plane++){
        quant_table[plane][0][0]        s   header_state
        for(level=0; level < spatial_decomposition_count; level++){
            quant_table[plane][level][1]s   header_state
@@ -131,8 +133,10 @@ block(level):

 residual:
    residual2(luma)
-    residual2(chroma_cr)
-    residual2(chroma_cb)
+    if (nb_planes > 2) {
+        residual2(chroma_cr)
+        residual2(chroma_cb)
+    }

 residual2:
    for(level=0; level<spatial_decomposition_count; level++){
@@ -146,7 +150,7 @@ residual2:
 subband:
    FIXME

-
+nb_plane_types = gray ? 1 : 2;

 Tag description:
 ----------------
@@ -168,7 +172,11 @@ spatial_decomposition_count
    FIXME

 colorspace_type
-    0
+    0   unspecified YcbCr
+    1   Gray
+    2   Gray + Alpha
+    3   GBR
+    4   GBRA
    this MUST NOT change within a bitstream

 chroma_h_shift
@@ -1,24 +0,0 @@
-Google Summer of Code and similar project guidelines
-
-Summer of Code is a project by Google in which students are paid to implement
-some nice new features for various participating open source projects ...
-
-This text is a collection of things to take care of for the next soc as
-it's a little late for this year's soc (2006).
-
-The Goal:
-Our goal in respect to soc is and must be of course exactly one thing and
-that is to improve FFmpeg, to reach this goal, code must
-* conform to the development policy and patch submission guidelines
-* must improve FFmpeg somehow (faster, smaller, "better",
-  more codecs supported, fewer bugs, cleaner, ...)
-
-for mentors and other developers to help students to reach that goal it is
-essential that changes to their codebase are publicly visible, clean and
-easy reviewable that again leads us to:
-* use of a revision control system like git
-* separation of cosmetic from non-cosmetic changes (this is almost entirely
-  ignored by mentors and students in soc 2006 which might lead to a surprise
-  when the code will be reviewed at the end before a possible inclusion in
-  FFmpeg, individual changes were generally not reviewable due to cosmetics).
-* frequent commits, so that comments can be provided early
@@ -1,258 +0,0 @@
-@chapter Syntax
-@c man begin SYNTAX
-
-This section documents the syntax and formats employed by the FFmpeg
-libraries and tools.
-
-@anchor{quoting_and_escaping}
-@section Quoting and escaping
-
-FFmpeg adopts the following quoting and escaping mechanism, unless
-explicitly specified. The following rules are applied:
-
-@itemize
-@item
-@code{'} and @code{\} are special characters (respectively used for
-quoting and escaping). In addition to them, there might be other
-special characters depending on the specific syntax where the escaping
-and quoting are employed.
-
-@item
-A special character is escaped by prefixing it with a '\'.
-
-@item
-All characters enclosed between '' are included literally in the
-parsed string. The quote character @code{'} itself cannot be quoted,
-so you may need to close the quote and escape it.
-
-@item
-Leading and trailing whitespaces, unless escaped or quoted, are
-removed from the parsed string.
-@end itemize
-
-Note that you may need to add a second level of escaping when using
-the command line or a script, which depends on the syntax of the
-adopted shell language.
-
-The function @code{av_get_token} defined in
-@file{libavutil/avstring.h} can be used to parse a token quoted or
-escaped according to the rules defined above.
-
-The tool @file{tools/ffescape} in the FFmpeg source tree can be used
-to automatically quote or escape a string in a script.
-
-@subsection Examples
-
-@itemize
-@item
-Escape the string @code{Crime d'Amour} containing the @code{'} special
-character:
-@example
-Crime d\'Amour
-@end example
-
-@item
-The string above contains a quote, so the @code{'} needs to be escaped
-when quoting it:
-@example
-'Crime d'\''Amour'
-@end example
-
-@item
-Include leading or trailing whitespaces using quoting:
-@example
-'  this string starts and ends with whitespaces  '
-@end example
-
-@item
-Escaping and quoting can be mixed together:
-@example
-' The string '\'string\'' is a string '
-@end example
-
-@item
-To include a literal @code{\} you can use either escaping or quoting:
-@example
-'c:\foo' can be written as c:\\foo
-@end example
-@end itemize
-
-@anchor{date syntax}
-@section Date
-
-The accepted syntax is:
-@example
-[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
-now
-@end example
-
-If the value is "now" it takes the current time.
-
-Time is local time unless Z is appended, in which case it is
-interpreted as UTC.
-If the year-month-day part is not specified it takes the current
-year-month-day.
-
-@anchor{time duration syntax}
-@section Time duration
-
-The accepted syntax is:
-@example
-[-][HH:]MM:SS[.m...]
-[-]S+[.m...]
-@end example
-
-@var{HH} expresses the number of hours, @var{MM} the number a of minutes
-and @var{SS} the number of seconds.
-
-@anchor{video size syntax}
-@section Video size
-Specify the size of the sourced video, it may be a string of the form
-@var{width}x@var{height}, or the name of a size abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item ntsc
-720x480
-@item pal
-720x576
-@item qntsc
-352x240
-@item qpal
-352x288
-@item sntsc
-640x480
-@item spal
-768x576
-@item film
-352x240
-@item ntsc-film
-352x240
-@item sqcif
-128x96
-@item qcif
-176x144
-@item cif
-352x288
-@item 4cif
-704x576
-@item 16cif
-1408x1152
-@item qqvga
-160x120
-@item qvga
-320x240
-@item vga
-640x480
-@item svga
-800x600
-@item xga
-1024x768
-@item uxga
-1600x1200
-@item qxga
-2048x1536
-@item sxga
-1280x1024
-@item qsxga
-2560x2048
-@item hsxga
-5120x4096
-@item wvga
-852x480
-@item wxga
-1366x768
-@item wsxga
-1600x1024
-@item wuxga
-1920x1200
-@item woxga
-2560x1600
-@item wqsxga
-3200x2048
-@item wquxga
-3840x2400
-@item whsxga
-6400x4096
-@item whuxga
-7680x4800
-@item cga
-320x200
-@item ega
-640x350
-@item hd480
-852x480
-@item hd720
-1280x720
-@item hd1080
-1920x1080
-@item 2k
-2048x1080
-@item 2kflat
-1998x1080
-@item 2kscope
-2048x858
-@item 4k
-4096x2160
-@item 4kflat
-3996x2160
-@item 4kscope
-4096x1716
-@end table
-
-@anchor{video rate syntax}
-@section Video rate
-
-Specify the frame rate of a video, expressed as the number of frames
-generated per second. It has to be a string in the format
-@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
-number or a valid video frame rate abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item ntsc
-30000/1001
-@item pal
-25/1
-@item qntsc
-30000/1001
-@item qpal
-25/1
-@item sntsc
-30000/1001
-@item spal
-25/1
-@item film
-24/1
-@item ntsc-film
-24000/1001
-@end table
-
-@anchor{ratio syntax}
-@section Ratio
-
-A ratio can be expressed as an expression, or in the form
-@var{numerator}:@var{denominator}.
-
-Note that a ratio with infinite (1/0) or negative value is
-considered valid, so you should check on the returned value if you
-want to exclude those values.
-
-The undefined value can be expressed using the "0:0" string.
-
-@anchor{color syntax}
-@section Color
-
-It can be the name of a color (case insensitive match) or a
-[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
-representing the alpha component.
-
-The alpha component may be a string composed by "0x" followed by an
-hexadecimal number or a decimal number between 0.0 and 1.0, which
-represents the opacity value (0x00/0.0 means completely transparent,
-0xff/1.0 completely opaque).
-If the alpha component is not specified then 0xff is assumed.
-
-The string "random" will result in a random color.
-
-@c man end SYNTAX
@@ -17,6 +17,7 @@ my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
 </head>
 <body>
 <div id="container">
+<div id="body">
 EOT

 $PRE_BODY_CLOSE = '</div></div>';
@@ -32,7 +33,7 @@ sub FFmpeg_print_page_foot($$)
        T2H_DEFAULT_program_string() : program_string();
    print $fh '<footer class="footer pagination-right">' . "\n";
    print $fh '<span class="label label-info">' . $program_string;
-    print $fh "</span></footer></div>\n";
+    print $fh "</span></footer></div></div></body>\n";
 }

 $float = \&FFmpeg_float;
@@ -92,8 +93,6 @@ $Texi2HTML::THISDOC{program_authors}

 $description
 <meta name="keywords" content="$longtitle">
-<meta name="resource-type" content="document">
-<meta name="distribution" content="global">
 <meta name="Generator" content="$Texi2HTML::THISDOC{program}">
 $encoding
 $CSS_LINES
@@ -1,4 +1,4 @@
-#! /usr/bin/perl
+#!/usr/bin/env perl

 #   Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.

@@ -121,7 +121,7 @@ INF: while(<$inf>) {
        $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);

        # start new chapter
-        $chapter_name = $1, push (@chapters_sequence, $chapter_name);
+        $chapter_name = $1, push (@chapters_sequence, $chapter_name) unless $skipping;
        $chapters{$chapter_name} = "" unless exists $chapters{$chapter_name};
        $chapter = "";
        $output = 1;
@@ -327,6 +327,9 @@ die "No filename or title\n" unless defined $fn && defined $tl;
 $chapters{NAME} = "$fn \- $tl\n";
 $chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};

+# always use utf8
+print "=encoding utf8\n\n";
+
 unshift @chapters_sequence, "NAME";
 for $chapter (@chapters_sequence) {
    if (exists $chapters{$chapter}) {
@@ -377,8 +380,8 @@ sub postprocess
    s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
    s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
    s/;\s+\@pxref\{(?:[^\}]*)\}//g;
-    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g;
-    s/\@ref\{([^\}]*)\}/$1/g;
+    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/B<$1>/g;
+    s/\@ref\{([^\}]*)\}/B<$1>/g;
    s/\@noindent\s*//g;
    s/\@refill//g;
    s/\@gol//g;
@@ -0,0 +1,32 @@
+#! /usr/bin/env perl
+
+# This script will print the dependency of a Texinfo file to stdout.
+# texidep.pl <src-path> <input.texi> <output.ext>
+
+use warnings;
+use strict;
+
+die unless @ARGV == 3;
+
+my ($src_path, $root, $target) = @ARGV;
+
+sub print_deps {
+    my ($file, $deps) = @_;
+    $deps->{$file} = 1;
+
+    open(my $fh, "<", "$file") or die "Cannot open file '$file': $!";
+    while (<$fh>) {
+        if (my ($i) = /^\@(?:verbatim)?include\s+(\S+)/) {
+            die "Circular dependency found in file $root\n" if exists $deps->{"doc/$1"};
+            print "$target: doc/$1\n";
+
+            # skip looking for config.texi dependencies, since it has
+            # none, and is not located in the source tree
+            if ("$1" ne "config.texi") {
+                print_deps("$src_path/doc/$1", {%$deps});
+            }
+        }
+    }
+}
+
+print_deps($root, {});
@@ -1,109 +0,0 @@
-This is a quick description of the viterbi aka dynamic programing
-algorthm.
-
-Its reason for existence is that wikipedia has become very poor on
-describing algorithms in a way that makes it useable for understanding
-them or anything else actually. It tends now to describe the very same
-algorithm under 50 different names and pages with few understandable
-by even people who fully understand the algorithm and the theory behind.
-
-Problem description: (that is what it can solve)
-assume we have a 2d table, or you could call it a graph or matrix if you
-prefer
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-
-That table has edges connecting points from each column to the next column
-and each edge has a score like: (only some edge and scores shown to keep it
-readable)
-
-
-    O--5--O-----O-----O-----O-----O
-     2   / 7   / \   / \   / \   /
-      \ /   \ /   \ /   \ /   \ /
-    O7-/--O--/--O--/--O--/--O--/--O
-     \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
-     /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
-    O3-/--O--/--O--/--O--/--O--/--O
-      / \   / \   / \   / \   / \
-     1   \ 9   \ /   \ /   \ /   \
-    O--2--O--1--O--5--O--3--O--8--O
-
-
-
-Our goal is to find a path from left to right through it which
-minimizes the sum of the score of all edges.
-(and of course left/right is just a convention here it could be top down too)
-Similarly the minimum could be the maximum by just fliping the sign,
-Example of a path with scores:
-
-    O   O   O   O   O   O   O
-
->---O.  O   O  .O-2-O   O   O
-      5.     .7      .
-    O   O-1-O   O   O 8 O   O
-                       .
-    O   O   O   O   O   O-1-O---> (sum here is 24)
-
-
-The viterbi algorthm now solves this simply column by column
-For the previous column each point has a best path and a associated
-score:
-
-    O-----5     O
-     \
-      \
-    O  \  1     O
-        \/
-        /\
-    O  /  2     O
-      /
-     /
-    O-----2     O
-
-
-To move one column forward we just need to find the best path and associated
-scores for the next column
-here are some edges we could choose from:
-
-
-    O-----5--3--O
-     \      \8
-      \       \
-    O  \  1--9--O
-        \/  \3
-        /\     \
-    O  /  2--1--O
-      /     \2
-     /        \
-    O-----2--4--O
-
-Finding the new best paths and scores for each point of our new column is
-trivial given we know the previous column best paths and scores:
-
-    O-----0-----8
-     \
-      \
-    O  \  0----10
-        \/
-        /\
-    O  /  0-----3
-      /     \
-     /        \
-    O     0     4
-
-
-the viterbi algorthm continues exactly like this column for column until the
-end and then just picks the path with the best score (above that would be the
-one with score 3)
-
-
-Author: Michael niedermayer
-Copyright LGPL
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .1.git
 .2.10