Merge remote-tracking branch 'qatar/release/0.6' into release/0.6

* qatar/release/0.6: Release notes and changelog for 0.6.4 Conflicts: Changelog Merged-by: Michael Niedermayer <michaelni@gmx.at>
Release notes and changelog for 0.6.4
2011-12-25 20:11:09 +01:00 · 2011-12-25 10:03:08 +01:00 · 2011-12-25 01:24:40 +01:00 · 2011-12-24 15:59:10 +01:00 · 2011-12-24 15:57:17 +01:00 · 2011-12-24 15:54:51 +01:00
5451 changed files with 287197 additions and 966133 deletions
--- a/.gitattributes
+++ b/.gitattributes
@@ -1 +0,0 @@
 *.pnm -diff -text
--- a/.gitignore
+++ b/.gitignore
@@ -1,91 +1,45 @@
-*.a
+.config
 .version
 *.o
 *.d
 *.def
 *.dll
 *.dylib
 *.exe
-*.exp
+*.ho
 *.gcda
 *.gcno
 *.h.c
 *.ilk
 *.lib
 *.pc
 *.pdb
 *.so
 *.so.*
 *.swp
 *.ver
 *-example
 *-test
 *_g
-/.config
+config.*
-/.version
+doc/*.1
-/ffmpeg
+doc/*.html
-/ffplay
+doc/*.pod
-/ffprobe
+doxy
-/ffserver
+ffmpeg
-/config.*
+ffplay
-/coverage.info
+ffprobe
-/doc/*.1
+ffserver
-/doc/*.3
+libavcodec/libavcodec*
-/doc/*.html
+libavcore/libavcore*
-/doc/*.pod
+libavdevice/libavdevice*
-/doc/config.texi
+libavfilter/libavfilter*
-/doc/avoptions_codec.texi
+libavformat/libavformat*
-/doc/avoptions_format.texi
+libavutil/avconfig.h
-/doc/doxy/html/
+libavutil/libavutil*
-/doc/examples/avio_list_dir
+libpostproc/libpostproc*
-/doc/examples/avio_reading
+libswscale/libswscale*
-/doc/examples/decoding_encoding
+tests/audiogen
-/doc/examples/demuxing_decoding
+tests/base64
-/doc/examples/extract_mvs
+tests/data
-/doc/examples/filter_audio
+tests/rotozoom
-/doc/examples/filtering_audio
+tests/seek_test
-/doc/examples/filtering_video
+tests/tiny_psnr
-/doc/examples/metadata
+tests/videogen
-/doc/examples/muxing
+tests/vsynth1
-/doc/examples/pc-uninstalled
+tests/vsynth2
-/doc/examples/remuxing
+tools/cws2fws
-/doc/examples/resampling_audio
+tools/graph2dot
-/doc/examples/scaling_video
+tools/lavfi-showfiltfmts
-/doc/examples/transcode_aac
+tools/pktdumper
-/doc/examples/transcoding
+tools/probetest
-/doc/fate.txt
+tools/qt-faststart
-/doc/print_options
+tools/trasher
-/lcov/
+tools/trasher*.d
-/libavcodec/*_tablegen
+version.h
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
 /tests/pixfmts.mak
 /tests/rotozoom
 /tests/test_copy.ffmeta
 /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
 /tools/probetest
 /tools/qt-faststart
 /tools/sidxindex
 /tools/trasher
 /tools/seek_print
 /tools/uncoded_frame
 /tools/zmqsend
--- a/COPYING.LGPLv2.1
+++ b/COPYING.LGPLv2.1
@@ -500,3 +500,5 @@ necessary.  Here is a sample; alter the names:
  Ty Coon, President of Vice
 That's all there is to it!
--- a/57
+++ b/57
@@ -1,6 +1,53 @@
-See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
+This file contains the name of the people who have contributed to
-get the names of people who have contributed to FFmpeg.
+FFmpeg. The names are sorted alphabetically by last name.
-To check the log, you can type the command "git log" in the FFmpeg
+Dénes Balatoni
-source directory, or browse the online repository at
+Michel Bardiaux
-http://source.ffmpeg.org.
+Fabrice Bellard
 Patrice Bensoussan
 Alex Beregszaszi
 BERO
 Thilo Borgmann
 Mario Brito
 Ronald Bultje
 Alex Converse
 Maarten Daniels
 Reimar Doeffinger
 Tim Ferguson
 Brian Foley
 Arpad Gereoffy
 Philip Gladstone
 Vladimir Gneushev
 Roine Gustafsson
 David Hammerton
 Wolfgang Hesseler
 Marc Hoffman
 Falk Hueffner
 Aurélien Jacobs
 Steven Johnson
 Zdenek Kabelac
 Robin Kay
 Todd Kirby
 Nick Kurshev
 Benjamin Larsson
 Loïc Le Loarer
 Daniel Maas
 Mike Melanson
 Loren Merritt
 Jeff Muizelaar
 Michael Niedermayer
 François Revol
 Peter Ross
 Måns Rullgård
 Stefano Sabatini
 Roman Shaposhnik
 Oded Shimon
 Dieter Shirley
 Konstantin Shishkov
 Juan J. Sierralta
 Ewald Snel
 Sascha Sommer
 Leon van Stuivenberg
 Roberto Togni
 Lionel Ulmer
 Reynaldo Verdejo
--- a/1193
+++ b/1193
--- a/1037
+++ b/1037
--- a/11
+++ b/11
@@ -0,0 +1,11 @@
 1) Type './configure' to create the configuration. A list of configure
 options is printed by running 'configure --help'.
 'configure' can be launched from a directory different from the FFmpeg
 sources to build the objects out of tree. To do this, use an absolute
 path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
 2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
 3) Type 'make install' to install all binaries and libraries you built.
--- a/INSTALL.md
+++ b/INSTALL.md
@@ -1,17 +0,0 @@
 #Installing FFmpeg:
 1. Type `./configure` to create the configuration. A list of configure
 options is printed by running `configure --help`.
    `configure` can be launched from a directory different from the FFmpeg
 sources to build the objects out of tree. To do this, use an absolute
 path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
 2. Then type `make` to build FFmpeg. GNU Make 3.81 or later is required.
 3. Type `make install` to install all binaries and libraries you built.
 NOTICE
 ------
 - Non system dependencies (e.g. libx264, libvpx) are disabled by default.
--- a/50
+++ b/50
@@ -0,0 +1,50 @@
 FFmpeg:
 -------
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
 or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
 files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
 FFmpeg.
 Some optional parts of FFmpeg are licensed under the GNU General Public License
 version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
 these parts are used by default, you have to explicitly pass --enable-gpl to
 configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are
 - libpostproc
 - optional MMX optimizations for YUV to RGB colorspace conversion in
  libswscale/x86/yuv2rgb_template.c
 - optional x86 optimizations in the files
  libavcodec/x86/h264_deblock_sse2.asm
  libavcodec/x86/h264_idct_sse2.asm
  libavcodec/x86/idct_mmx.c
 - the X11 grabber in libavdevice/x11grab.c
 There are a handful of files under other licensing terms, namely:
 * The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
  are taken from libjpeg, see the top of the files for licensing details.
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
 the configure parameter --enable-version3 will activate this licensing option
 for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
 COPYING.GPLv3 to learn the exact legal terms that apply in this case.
 external libraries:
 -------------------
 Some external libraries, e.g. libx264, are under GPL and can be used in
 conjunction with FFmpeg. They require --enable-gpl to be passed to configure
 as well.
 The OpenCORE external libraries are under the Apache License 2.0. That license
 is incompatible with the LGPL v2.1 and the GPL v2, but not with version 3 of
 those licenses. So to combine the OpenCORE libraries with FFmpeg, the license
 version needs to be upgraded by passing --enable-version3 to configure.
 The nonfree external library libfaac can be hooked up in FFmpeg. You need to
 pass --enable-nonfree to configure to enable it. Employ this option with care
 as FFmpeg then becomes nonfree and unredistributable.
--- a/LICENSE.md
+++ b/LICENSE.md
@@ -1,112 +0,0 @@
 #FFmpeg:
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
 or later (LGPL v2.1+). Read the file `COPYING.LGPLv2.1` for details. Some other
 files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
 FFmpeg.
 Some optional parts of FFmpeg are licensed under the GNU General Public License
 version 2 or later (GPL v2+). See the file `COPYING.GPLv2` for details. None of
 these parts are used by default, you have to explicitly pass `--enable-gpl` to
 configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are:
 - libpostproc
 - optional x86 optimizations in the files
  - `libavcodec/x86/flac_dsp_gpl.asm`
  - `libavcodec/x86/idct_mmx.c`
 - libutvideo encoding/decoding wrappers in
  `libavcodec/libutvideo*.cpp`
 - the X11 grabber in `libavdevice/x11grab.c`
 - the swresample test app in
  `libswresample/swresample-test.c`
 - the `texi2pod.pl` tool
 - the following filters in libavfilter:
    - `f_ebur128.c`
    - `vf_blackframe.c`
    - `vf_boxblur.c`
    - `vf_colormatrix.c`
    - `vf_cover_rect.c`
    - `vf_cropdetect.c`
    - `vf_delogo.c`
    - `vf_eq.c`
    - `vf_find_rect.c`
    - `vf_fspp.c`
    - `vf_geq.c`
    - `vf_histeq.c`
    - `vf_hqdn3d.c`
    - `vf_interlace.c`
    - `vf_kerndeint.c`
    - `vf_mcdeint.c`
    - `vf_mpdecimate.c`
    - `vf_owdenoise.c`
    - `vf_perspective.c`
    - `vf_phase.c`
    - `vf_pp.c`
    - `vf_pp7.c`
    - `vf_pullup.c`
    - `vf_sab.c`
    - `vf_smartblur.c`
    - `vf_repeatfields.c`
    - `vf_spp.c`
    - `vf_stereo3d.c`
    - `vf_super2xsai.c`
    - `vf_tinterlace.c`
    - `vf_uspp.c`
    - `vsrc_mptestsrc.c`
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
 the configure parameter `--enable-version3` will activate this licensing option
 for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
 `COPYING.GPLv3` to learn the exact legal terms that apply in this case.
 There are a handful of files under other licensing terms, namely:
 * The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
  `libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
  licensing details. Specifically note that you must credit the IJG in the
  documentation accompanying your program if you only distribute executables.
  You must also indicate any changes including additions and deletions to
  those three files in the documentation.
 * `tests/reference.pnm` is under the expat license.
 external libraries
 ==================
 FFmpeg can be combined with a number of external libraries, which sometimes
 affect the licensing of binaries resulting from the combination.
 compatible libraries
 --------------------
 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
 license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 version 3 of those licenses. So to combine these libraries with FFmpeg, the
 license version needs to be upgraded by passing `--enable-version3` to configure.
 incompatible libraries
 ----------------------
 The Fraunhofer AAC library, FAAC and aacplus are under licenses which
 are incompatible with the GPLv2 and v3. We do not know for certain if their
 licenses are compatible with the LGPL.
 If you wish to enable these libraries, pass `--enable-nonfree` to configure.
 But note that if you enable any of these libraries the resulting binary will
 be under a complex license mix that is more restrictive than the LGPL and that
 may result in additional obligations. It is possible that these
 restrictions cause the resulting binary to be unredistributeable.
--- a/323
+++ b/323
@@ -4,16 +4,11 @@ FFmpeg maintainers
 Below is a list of the people maintaining different parts of the
 FFmpeg code.
 Please try to keep entries where you are the maintainer up to date!
 Names in () mean that the maintainer currently has no time to maintain the code.
 A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
 patches and related discussions.
 Project Leader
 ==============
 Michael Niedermayer
  final design decisions
@@ -24,13 +19,10 @@ ffmpeg:
  ffmpeg.c                              Michael Niedermayer
 ffplay:
-  ffplay.c                              Marton Balint
+  ffplay.c                              Michael Niedermayer
 ffprobe:
  ffprobe.c                             Stefano Sabatini
 ffserver:
-  ffserver.c                            Reynaldo H. Verdejo Pinochet
+  ffserver.c, ffserver.h                Baptiste Coudurier
 Commandline utility code:
  cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -42,24 +34,14 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
+documentation                           Mike Melanson, Diego Biurrun
-build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
+website                                 Robert Swain
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Lou Logan
+build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
 project server                          Diego Biurrun, Mans Rullgard
 mailinglists                            Michael Niedermayer, Baptiste Coudurier
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
-release management                      Michael Niedermayer
+release management                      Diego Biurrun, Reinhard Tartler
 Communication
 =============
 website                                 Deby Barbara Lepage
 fate.ffmpeg.org                         Timothy Gu
 Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
 mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
 Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
 Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu
 libavutil
@@ -71,24 +53,11 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer
 Other:
  bprint                                Nicolas George
  bswap.h
  des                                   Reimar Doeffinger
  dynarray.h                            Nicolas George
  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
+  mathematics.c, mathematics.h          Michael Niedermayer
-  ripemd.c, ripemd.h                    James Almer
+  integer.c, integer.h                  Michael Niedermayer
-  timecode                              Clément Bœsch
+  bswap.h
 libavcodec
@@ -99,14 +68,16 @@ 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:
    bitstream.c, bitstream.h            Michael Niedermayer
  CABAC:
    cabac.h, cabac.c                    Michael Niedermayer
  codec names:
    codec_names.sh                      Nicolas George
  DSP utilities:
    dsputils.c, dsputils.h              Michael Niedermayer
  entropy coding:
@@ -127,110 +98,77 @@ Generic Parts:
    simple_idct.c, simple_idct.h        Michael Niedermayer
  postprocessing:
    libpostproc/*                       Michael Niedermayer
  table generation:
    tableprint.c, tableprint.h          Reimar Doeffinger
  fixed point FFT:
    fft*                                Zeljko Lukac
  Text Subtitles                        Clément Bœsch
 Codecs:
  4xm.c                                 Michael Niedermayer
  8bps.c                                Roberto Togni
  8svx.c                                Jaikrishnan Menon
  aasc.c                                Kostya Shishkov
  aac*, sbr.h                           Alex Converse
  ac3*                                  Justin Ruggles
  alacenc.c                             Jaikrishnan Menon
  alsdec.c                              Thilo Borgmann
  apedec.c                              Kostya Shishkov
  ass*                                  Aurelien Jacobs
  asv*                                  Michael Niedermayer
  atrac3*                               Benjamin Larsson
  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
  cdxl.c                                Paul B Mahol
  celp_filters.*                        Vitor Sessak
  cinepak.c                             Roberto Togni
  cinepakenc.c                          Rl / Aetey G.T. AB
  ccaption_dec.c                        Anshul Maheshwari
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
  cook.c, cookdata.h                    Benjamin Larsson
  cpia.c                                Stephan Hilb
  crystalhd.c                           Philip Langdale
  cscd.c                                Reimar Doeffinger
  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
  dss_sp.c                              Oleksij Rempel, Michael Niedermayer
  dv.c                                  Roman Shaposhnik
  dvbsubdec.c                           Anshul Maheshwari
  dxa.c                                 Kostya Shishkov
  dv.c                                  Roman Shaposhnik
  eacmv*, eaidct*, eat*                 Peter Ross
-  exif.c, exif.h                        Thilo Borgmann
+  ffv1.c                                Michael Niedermayer
  ffv1*                                 Michael Niedermayer
  ffwavesynth.c                         Nicolas George
  fic.c                                 Derek Buitenhuis
  flac*                                 Justin Ruggles
  flashsv*                              Benjamin Larsson
  flicvideo.c                           Mike Melanson
  g722.c                                Martin Storsjo
  g726.c                                Roman Shaposhnik
  gifdec.c                              Baptiste Coudurier
  h264*                                 Loren Merritt, Michael Niedermayer
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
-  h264*                                 Loren Merritt, Michael Niedermayer
+  huffyuv.c                             Michael Niedermayer
  huffyuv*                              Michael Niedermayer, Christophe Gisquet
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
  indeo2*                               Kostya Shishkov
  indeo5*                               Kostya Shishkov
  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
  libdirac*                             David Conrad
  libgsm.c                              Michel Bardiaux
  libdirac*                             David Conrad
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
  libspeexdec.c                         Justin Ruggles
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
  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
  mimic.c                               Ramiro Polla
-  mjpeg*.c                              Michael Niedermayer
+  mjpeg.c                               Michael Niedermayer
  mlp*                                  Ramiro Polla
  mmvideo.c                             Peter Ross
  mpc*                                  Kostya Shishkov
  mpeg12.c, mpeg12data.h                Michael Niedermayer
  mpegvideo.c, mpegvideo.h              Michael Niedermayer
  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
  msvideo1.c                            Mike Melanson
  nellymoserdec.c                       Benjamin Larsson
  nuv.c                                 Reimar Doeffinger
  nvenc.c                               Timo Rothenpieler
  paf.*                                 Paul B Mahol
  pcx.c                                 Ivo van Poorten
  pgssubdec.c                           Reimar Doeffinger
  ptx.c                                 Ivo van Poorten
  qcelp*                                Reynaldo H. Verdejo Pinochet
  qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
@@ -244,17 +182,14 @@ Codecs:
  rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
  rv10.c                                Michael Niedermayer
  rv3*                                  Kostya Shishkov
-  rv4*                                  Kostya Shishkov, Christophe Gisquet
+  rv4*                                  Kostya Shishkov
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
-  smvjpegdec.c                          Ash Hughes
+  snow.c                                Michael Niedermayer, Loren Merritt
  snow*                                 Michael Niedermayer, Loren Merritt
  sonic.c                               Alex Beregszaszi
  srt*                                  Aurelien Jacobs
  sunrast.c                             Ivo van Poorten
  svq3.c                                Michael Niedermayer
  tak*                                  Paul B Mahol
  targa.c                               Kostya Shishkov
  tiff.c                                Kostya Shishkov
  truemotion1*                          Mike Melanson
@@ -262,24 +197,17 @@ 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
  vb.c                                  Kostya Shishkov
-  vble.c                                Derek Buitenhuis
+  vc1*                                  Kostya Shishkov
  vc1*                                  Kostya Shishkov, Christophe Gisquet
  vcr1.c                                Michael Niedermayer
  vda_h264_dec.c                        Xidorn Quan
  vima.c                                Paul B Mahol
  vmnc.c                                Kostya Shishkov
-  vorbisdec.c                           Denes Balatoni, David Conrad
+  vorbis_enc.c                          Oded Shimon
-  vorbisenc.c                           Oded Shimon
+  vorbis_dec.c                          Denes Balatoni
  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
@@ -287,20 +215,13 @@ Codecs:
  wmv2.c                                Michael Niedermayer
  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
  zerocodec.c                           Derek Buitenhuis
  zmbv*                                 Kostya Shishkov
 Hardware acceleration:
  crystalhd.c                           Philip Langdale
  dxva2*                                Laurent Aimar
  libstagefright.cpp                    Mohamed Naufal
  vaapi*                                Gwenole Beauchesne
  vda*                                  Sebastien Zwickert
  vdpau*                                Carl Eugen Hoyos
@@ -310,66 +231,10 @@ libavdevice
    libavdevice/avdevice.h
  avfoundation.m                        Thilo Borgmann
  decklink*                             Deti Fliegl
  dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
  fbdev_enc.c                           Lukasz Marek
  gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
  iec61883.c                            Georg Lippitsch
  lavfi                                 Stefano Sabatini
  libdc1394.c                           Roman Shaposhnik
-  opengl_enc.c                          Lukasz Marek
+  v4l2.c                                Luca Abeni
  pulse_audio_enc.c                     Lukasz Marek
  qtkit.m                               Thilo Borgmann
  sdl                                   Stefano Sabatini
  v4l2.c                                Giorgio Vazzana
  vfwcap.c                              Ramiro Polla
  xv.c                                  Lukasz Marek
 libavfilter
 ===========
 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
  af_silenceremove.c                    Paul B Mahol
  avf_avectorscope.c                    Paul B Mahol
  avf_showcqt.c                         Muhammad Faiz
  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_hqx.c                              Clément Bœsch
  vf_idet.c                             Pascal Massimino
  vf_il.c                               Paul B Mahol
  vf_lenscorrection.c                   Daniel Oberhoff
  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:
  vsrc_mandelbrot.c                     Michael Niedermayer
 libavformat
 ===========
@@ -384,142 +249,77 @@ Generic parts:
 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
  apngdec.c                             Benoit Fouet
  ass*                                  Aurelien Jacobs
  astdec.c                              Paul B Mahol
  astenc.c                              James Almer
  avi*                                  Michael Niedermayer
  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
  avr.c                                 Paul B Mahol
  bink.c                                Peter Ross
  brstm.c                               Paul B Mahol
  caf*                                  Peter Ross
  cdxl.c                                Paul B Mahol
  crc.c                                 Michael Niedermayer
  daud.c                                Reimar Doeffinger
  dss.c                                 Oleksij Rempel, Michael Niedermayer
  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
  electronicarts.c                      Peter Ross
  epafdec.c                             Paul B Mahol
  ffm*                                  Baptiste Coudurier
  flac*                                 Justin Ruggles
  flic.c                                Mike Melanson
  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
+  idroq.c                               Mike Melanson
  iff.c                                 Jaikrishnan Menon
  img2*.c                               Michael Niedermayer
  ipmovie.c                             Mike Melanson
-  ircam*                                Paul B Mahol
+  img2.c                                Michael Niedermayer
  iss.c                                 Stefan Gehrer
  jacosub*                              Clément Bœsch
  jvdec.c                               Peter Ross
  libmodplug.c                          Clément Bœsch
  libnut.c                              Oded Shimon
  lmlm4.c                               Ivo van Poorten
  lvfdec.c                              Paul B Mahol
  lxfdec.c                              Tomas Härdin
  matroska.c                            Aurelien Jacobs
  matroskadec.c                         Aurelien Jacobs
  matroskaenc.c                         David Conrad
  matroska subtitles (matroskaenc.c)    John Peebles
  metadata*                             Aurelien Jacobs
  mgsts.c                               Paul B Mahol
  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
  mov.c                                 Michael Niedermayer, Baptiste Coudurier
-  movenc.c                              Baptiste Coudurier, Matthieu Bouron
+  movenc.c                              Michael Niedermayer, Baptiste Coudurier
  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
  mpegenc.c                             Michael Niedermayer
-  mpegts.c                              Marton Balint
+  mpegts*                               Baptiste Coudurier
  mpegtsenc.c                           Baptiste Coudurier
  msnwc_tcp.c                           Ramiro Polla
  mtv.c                                 Reynaldo H. Verdejo Pinochet
  mxf*                                  Baptiste Coudurier
  mxfdec.c                              Tomas Härdin
  nistspheredec.c                       Paul B Mahol
  nsvdec.c                              Francois Revol
-  nut*                                  Michael Niedermayer
+  nut.c                                 Michael Niedermayer
  nuv.c                                 Reimar Doeffinger
  oggdec.c, oggdec.h                    David Conrad
  oggenc.c                              Baptiste Coudurier
  oggparse*.c                           David Conrad
  oma.c                                 Maxim Poliakovski
  paf.c                                 Paul B Mahol
  psxstr.c                              Mike Melanson
  pva.c                                 Ivo van Poorten
  pvfdec.c                              Paul B Mahol
  r3d.c                                 Baptiste Coudurier
  raw.c                                 Michael Niedermayer
  rdt.c                                 Ronald S. Bultje
  rl2.c                                 Sascha Sommer
  rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
  rtmp*                                 Kostya Shishkov
-  rtp.c, rtpenc.c                       Martin Storsjo
+  rtp.c, rtpenc.c                       Luca Abeni
-  rtpdec_ac3.*                          Gilles Chanteperdrix
+  rtp_asf.*                             Ronald S. Bultje
-  rtpdec_dv.*                           Thomas Volkert
+  rtp_mpv.*, rtp_aac.*                  Luca Abeni
  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
  rtpdec_hevc.*, rtpenc_hevc.*          Thomas Volkert
  rtpdec_mpa_robust.*                   Gilles Chanteperdrix
  rtpdec_asf.*                          Ronald S. Bultje
  rtpdec_vp9.c                          Thomas Volkert
  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
  rtsp.c                                Luca Barbato
-  sbgdec.c                              Nicolas George
+  sdp.c                                 Luca Abeni
  sdp.c                                 Martin Storsjo
  segafilm.c                            Mike Melanson
  segment.c                             Stefano Sabatini
  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
  tta.c                                 Alex Beregszaszi
  txd.c                                 Ivo van Poorten
  voc.c                                 Aurelien Jacobs
  wav.c                                 Michael Niedermayer
  wc3movie.c                            Mike Melanson
  webm dash (matroskaenc.c)             Vignesh Venkatasubramanian
  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
  wv.c                                  Kostya Shishkov
  wvenc.c                               Paul B Mahol
 Protocols:
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
  libssh.c                              Lukasz Marek
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni
  icecast.c                             Marvin Scholz
 libswresample
 =============
 Generic parts:
  audioconvert.c                        Michael Niedermayer
  dither.c                              Michael Niedermayer
  rematrix*.c                           Michael Niedermayer
  swresample*.c                         Michael Niedermayer
 Resamplers:
  resample*.c                           Michael Niedermayer
  soxr_resample.c                       Rob Sykes
 Operating systems / CPU architectures
@@ -528,65 +328,32 @@ Operating systems / CPU architectures
 Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
-MIPS                                    Mans Rullgard, Nedeljko Babic
+MIPS                                    Mans Rullgard
 BeOS                                    Francois Revol
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
 Windows MSVC                            Matthew Oliver
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer
-Releases
+GnuPG Fingerprints of maintainers and others who have svn write access
-========
+======================================================================
 2.7                                     Michael Niedermayer
 2.6                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
 2.4                                     Michael Niedermayer
 2.2                                     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
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
 Clément Bœsch                 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
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
+Panagiotis Issaris            515C E262 10A8 FDCE 5481 7B9C 3AD7 D9A5 071D B3A9
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
-Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
+Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
 Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
 Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
--- a/406
+++ b/406
@@ -1,87 +1,62 @@
 MAIN_MAKEFILE=1
 include config.mak
-vpath %.c    $(SRC_PATH)
+SRC_DIR = $(SRC_PATH_BARE)
 vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
 vpath %.m    $(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 $(SRC_PATH)
-AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
+vpath %.texi $(SRC_PATH_BARE)
 AVPROGS-$(CONFIG_FFPLAY)   += ffplay
 AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
 AVPROGS-$(CONFIG_FFSERVER) += ffserver
-AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS-$(CONFIG_FFMPEG)   += ffmpeg
-INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS-$(CONFIG_FFPLAY)   += ffplay
-PROGS      += $(AVPROGS)
+PROGS-$(CONFIG_FFPROBE)  += ffprobe
 PROGS-$(CONFIG_FFSERVER) += ffserver
-AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
+PROGS      := $(addsuffix   $(EXESUF), $(PROGS-yes))
-ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
+PROGS_G     = $(addsuffix _g$(EXESUF), $(PROGS-yes))
-ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+OBJS        = $(addsuffix .o,          $(PROGS-yes)) cmdutils.o
 MANPAGES    = $(addprefix doc/, $(addsuffix .1, $(PROGS-yes)))
 TOOLS       = $(addprefix tools/, $(addsuffix $(EXESUF), cws2fws pktdumper probetest qt-faststart trasher))
 HOSTPROGS   = $(addprefix tests/, audiogen videogen rotozoom tiny_psnr)
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
+BASENAMES   = ffmpeg ffplay ffprobe ffserver
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
+ALLPROGS    = $(addsuffix   $(EXESUF), $(BASENAMES))
 ALLPROGS_G  = $(addsuffix _g$(EXESUF), $(BASENAMES))
 ALLMANPAGES = $(addsuffix .1, $(BASENAMES))
-OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
+FFLIBS-$(CONFIG_AVDEVICE) += avdevice
-OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
+FFLIBS-$(CONFIG_AVFILTER) += avfilter
-OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
+FFLIBS-$(CONFIG_AVFORMAT) += avformat
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
+FFLIBS-$(CONFIG_AVCODEC)  += avcodec
-OBJS-ffserver                 += ffserver_config.o
+FFLIBS-$(CONFIG_POSTPROC) += postproc
-
+FFLIBS-$(CONFIG_SWSCALE)  += swscale
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 TOOLS       = qt-faststart trasher uncoded_frame
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 # $(FFLIBS-yes) needs to be in linking order
 FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
 FFLIBS-$(CONFIG_AVFILTER)   += avfilter
 FFLIBS-$(CONFIG_AVFORMAT)   += avformat
 FFLIBS-$(CONFIG_AVCODEC)    += avcodec
 FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_POSTPROC)   += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
 FFLIBS-$(CONFIG_SWSCALE)    += swscale
 FFLIBS := avutil
-DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
+DATA_FILES := $(wildcard $(SRC_DIR)/ffpresets/*.ffpreset)
 EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
-SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
+SKIPHEADERS = cmdutils_common_opts.h
-include $(SRC_PATH)/common.mak
+include common.mak
 FF_LDFLAGS   := $(FFLDFLAGS)
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
-all: $(AVPROGS)
+ALL_TARGETS-$(CONFIG_DOC)       += documentation
-$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
+ifdef PROGS
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
+INSTALL_TARGETS-yes             += install-progs install-data
 INSTALL_TARGETS-$(CONFIG_DOC)   += install-man
 endif
 INSTALL_PROGS_TARGETS-$(CONFIG_SHARED) = install-libs
-tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
+all: $(FF_DEP_LIBS) $(PROGS) $(ALL_TARGETS-yes)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
 tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
-config.h: .config
+$(PROGS): %$(EXESUF): %_g$(EXESUF)
-.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
+	$(CP) $< $@
-	@-tput bold 2>/dev/null
+	$(STRIP) $@
 	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
-SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
+SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
-               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
+               ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
-               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
+               HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS
               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MSA-OBJS       \
               LOONGSON3-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
 define RESET
 $(1) :=
@@ -91,109 +66,288 @@ endef
 define DOSUBDIR
 $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
-include $(SRC_PATH)/$(1)/Makefile
+include $(1)/Makefile
 -include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
 -include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
 include $(SRC_PATH)/library.mak
 endef
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
-include $(SRC_PATH)/doc/Makefile
+ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
 ffserver_g$(EXESUF): FF_LDFLAGS += $(FFSERVERLDFLAGS)
-define DOPROG
+%_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
-OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
+	$(LD) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef
-$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
+tools/%$(EXESUF): tools/%.o
 	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
-ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
+tools/%.o: tools/%.c
 	$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_O) $<
-$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+ffplay.o ffplay.d: CFLAGS += $(SDL_CFLAGS)
 	$(CP) $< $@
 	$(STRIP) $@
-%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
+VERSION_SH  = $(SRC_PATH_BARE)/version.sh
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+GIT_LOG     = $(SRC_PATH_BARE)/.git/logs/HEAD
 SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
-OBJDIRS += tools
+.version: $(wildcard $(GIT_LOG) $(SVN_ENTRIES)) $(VERSION_SH) config.mak
 -include $(wildcard tools/*.d)
 VERSION_SH  = $(SRC_PATH)/version.sh
 GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
-libavutil/ffversion.h .version:
+version.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
+	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
 	$(Q)touch .version
 # force version.sh to run whenever version might have changed
 -include .version
-ifdef AVPROGS
+alltools: $(TOOLS)
 install: install-progs install-data
 endif
-install: install-libs install-headers
+documentation: $(addprefix doc/, developer.html faq.html ffmpeg-doc.html \
                                 ffplay-doc.html ffprobe-doc.html ffserver-doc.html \
                                 general.html libavfilter.html $(ALLMANPAGES))
-install-libs: install-libs-yes
+doc/%.html: TAG = HTML
 doc/%.html: doc/%.texi
 	$(M)cd doc && texi2html -monolithic -number $(<:doc/%=%)
-install-progs-yes:
+doc/%.pod: TAG = POD
-install-progs-$(CONFIG_SHARED): install-libs
+doc/%.pod: doc/%-doc.texi
 	$(M)doc/texi2pod.pl $< $@
-install-progs: install-progs-yes $(AVPROGS)
+doc/%.1: TAG = MAN
 doc/%.1: doc/%.pod
 	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
 install: $(INSTALL_TARGETS-yes)
 install-progs: $(PROGS) $(INSTALL_PROGS_TARGETS-yes)
 	$(Q)mkdir -p "$(BINDIR)"
-	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
+	$(INSTALL) -c -m 755 $(PROGS) "$(BINDIR)"
-install-data: $(DATA_FILES) $(EXAMPLES_FILES)
+install-data: $(DATA_FILES)
-	$(Q)mkdir -p "$(DATADIR)/examples"
+	$(Q)mkdir -p "$(DATADIR)"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
 	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
-uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
+install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
 	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
 uninstall: uninstall-progs uninstall-data uninstall-man
 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
-clean::
+uninstall-man:
-	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
 testclean:
 	$(RM) -r tests/vsynth1 tests/vsynth2 tests/data
 	$(RM) $(addprefix tests/,$(CLEANSUFFIXES))
 	$(RM) tests/seek_test$(EXESUF) tests/seek_test.o
 	$(RM) $(addprefix tests/,$(addsuffix $(HOSTEXESUF),audiogen videogen rotozoom tiny_psnr))
 clean:: testclean
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(CLEANSUFFIXES:%=tools/%)
+	$(RM) doc/*.html doc/*.pod doc/*.1
-	$(RM) -r coverage-html
+	$(RM) $(TOOLS)
 	$(RM) -rf coverage.info lcov
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) version.h config.* libavutil/avconfig.h
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
-check: all alltools examples testprogs fate
+# regression tests
-include $(SRC_PATH)/tests/Makefile
+check: test checkheaders
-$(sort $(OBJDIRS)):
+fulltest test: codectest lavftest seektest
 	$(Q)mkdir -p $@
-# Dummy rule to stop make trying to rebuild removed or renamed headers
+FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
-%.h:
+SEEK_REFFILE     = $(SRC_PATH)/tests/seek.regression.ref
 	@:
-# Disable suffix rules.  Most of the builtin rules are suffix rules,
+ENCDEC = $(and $(CONFIG_$(1)_ENCODER),$(CONFIG_$(1)_DECODER))
-# so this saves some time on slow systems.
+MUXDEM = $(and $(CONFIG_$(1)_MUXER),$(CONFIG_$(or $(2),$(1))_DEMUXER))
 .SUFFIXES:
-.PHONY: all all-yes alltools check *clean config install*
+VCODEC_TESTS =
-.PHONY: testprogs uninstall*
+VCODEC_TESTS-$(call ENCDEC,ASV1)             += asv1
 VCODEC_TESTS-$(call ENCDEC,ASV2)             += asv2
 VCODEC_TESTS-$(call ENCDEC,DNXHD)            += dnxhd_1080i dnxhd_720p dnxhd_720p_rd
 VCODEC_TESTS-$(call ENCDEC,DVVIDEO)          += dv dv50
 VCODEC_TESTS-$(call ENCDEC,FFV1)             += ffv1
 VCODEC_TESTS-$(call ENCDEC,FLASHSV)          += flashsv
 VCODEC_TESTS-$(call ENCDEC,FLV)              += flv
 VCODEC_TESTS-$(call ENCDEC,H261)             += h261
 VCODEC_TESTS-$(call ENCDEC,H263)             += h263 h263p
 VCODEC_TESTS-$(call ENCDEC,HUFFYUV)          += huffyuv
 VCODEC_TESTS-$(call ENCDEC,JPEGLS)           += jpegls
 VCODEC_TESTS-$(call ENCDEC,MJPEG)            += mjpeg ljpeg
 VCODEC_TESTS-$(call ENCDEC,MPEG1VIDEO)       += mpeg mpeg1b
 VCODEC_TESTS-$(call ENCDEC,MPEG2VIDEO)       += mpeg2 mpeg2thread
 VCODEC_TESTS-$(call ENCDEC,MPEG4)            += mpeg4 mpeg4adv mpeg4nr mpeg4thread error rc
 VCODEC_TESTS-$(call ENCDEC,MSMPEG4V1)        += msmpeg4
 VCODEC_TESTS-$(call ENCDEC,MSMPEG4V2)        += msmpeg4v2
 VCODEC_TESTS-$(call ENCDEC,ROQ)              += roq
 VCODEC_TESTS-$(call ENCDEC,RV10)             += rv10
 VCODEC_TESTS-$(call ENCDEC,RV20)             += rv20
 VCODEC_TESTS-$(call ENCDEC,SNOW)             += snow snowll
 VCODEC_TESTS-$(call ENCDEC,SVQ1)             += svq1
 VCODEC_TESTS-$(call ENCDEC,WMV1)             += wmv1
 VCODEC_TESTS-$(call ENCDEC,WMV2)             += wmv2
 ACODEC_TESTS =
 ACODEC_TESTS-$(call ENCDEC,AC3)              += ac3
 ACODEC_TESTS-$(call ENCDEC,ADPCM_G726)       += g726
 ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_QT)     += adpcm_ima_qt
 ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_WAV)    += adpcm_ima_wav
 ACODEC_TESTS-$(call ENCDEC,ADPCM_MS)         += adpcm_ms
 ACODEC_TESTS-$(call ENCDEC,ADPCM_SWF)        += adpcm_swf
 ACODEC_TESTS-$(call ENCDEC,ADPCM_YAMAHA)     += adpcm_yam
 ACODEC_TESTS-$(call ENCDEC,ALAC)             += alac
 ACODEC_TESTS-$(call ENCDEC,FLAC)             += flac
 ACODEC_TESTS-$(call ENCDEC,MP2)              += mp2
 ACODEC_TESTS-$(call ENCDEC,PCM_S16LE)        += pcm         # fixme
 ACODEC_TESTS-$(call ENCDEC,WMAV1)            += wmav1
 ACODEC_TESTS-$(call ENCDEC,WMAV1)            += wmav2
 LAVF_TESTS =
 LAVF_TESTS-$(call MUXDEM,AIFF)               += aiff
 LAVF_TESTS-$(call MUXDEM,PCM_ALAW)           += alaw
 LAVF_TESTS-$(call MUXDEM,ASF)                += asf
 LAVF_TESTS-$(call MUXDEM,AU)                 += au
 LAVF_TESTS-$(call MUXDEM,AVI)                += avi
 LAVF_TESTS-$(call ENCDEC,BMP)                += bmp
 LAVF_TESTS-$(call MUXDEM,DV)                 += dv_fmt
 LAVF_TESTS-$(call MUXDEM,FFM)                += ffm
 LAVF_TESTS-$(call MUXDEM,FLV)                += flv_fmt
 LAVF_TESTS-$(call ENCDEC,GIF)                += gif
 LAVF_TESTS-$(call MUXDEM,GXF)                += gxf
 LAVF_TESTS-$(call ENCDEC,MJPEG)              += jpg
 LAVF_TESTS-$(call MUXDEM,MATROSKA)           += mkv
 LAVF_TESTS-$(call MUXDEM,MMF)                += mmf
 LAVF_TESTS-$(call MUXDEM,MOV)                += mov
 LAVF_TESTS-$(call MUXDEM,MPEG1SYSTEM,MPEGPS) += mpg
 LAVF_TESTS-$(call MUXDEM,PCM_MULAW)          += mulaw
 LAVF_TESTS-$(call MUXDEM,MXF)                += mxf
 LAVF_TESTS-$(call MUXDEM,NUT)                += nut
 LAVF_TESTS-$(call MUXDEM,OGG)                += ogg
 LAVF_TESTS-$(call ENCDEC,PBM)                += pbmpipe
 LAVF_TESTS-$(call ENCDEC,PCX)                += pcx
 LAVF_TESTS-$(call ENCDEC,PGM)                += pgm pgmpipe
 LAVF_TESTS-$(call MUXDEM,RAWVIDEO)           += pixfmt
 LAVF_TESTS-$(call ENCDEC,PPM)                += ppm ppmpipe
 LAVF_TESTS-$(call MUXDEM,RM)                 += rm
 LAVF_TESTS-$(call ENCDEC,SGI)                += sgi
 LAVF_TESTS-$(call MUXDEM,SWF)                += swf
 LAVF_TESTS-$(call ENCDEC,TARGA)              += tga
 LAVF_TESTS-$(call ENCDEC,TIFF)               += tiff
 LAVF_TESTS-$(call MUXDEM,MPEGTS)             += ts
 LAVF_TESTS-$(call MUXDEM,VOC)                += voc
 LAVF_TESTS-$(call MUXDEM,WAV)                += wav
 LAVF_TESTS-$(call MUXDEM,YUV4MPEGPIPE)       += yuv4mpeg
 LAVFI_TESTS =           \
    crop                \
    crop_scale          \
    crop_scale_vflip    \
    crop_vflip          \
    null                \
    scale200            \
    scale500            \
    vflip               \
    vflip_crop          \
    vflip_vflip         \
 ACODEC_TESTS := $(addprefix regtest-, $(ACODEC_TESTS) $(ACODEC_TESTS-yes))
 VCODEC_TESTS := $(addprefix regtest-, $(VCODEC_TESTS) $(VCODEC_TESTS-yes))
 LAVF_TESTS  := $(addprefix regtest-, $(LAVF_TESTS)  $(LAVF_TESTS-yes))
 LAVFI_TESTS := $(addprefix regtest-, $(LAVFI_TESTS) $(LAVFI_TESTS-yes))
 CODEC_TESTS = $(VCODEC_TESTS) $(ACODEC_TESTS)
 codectest: $(CODEC_TESTS)
 lavftest:  $(LAVF_TESTS)
 lavfitest: $(LAVFI_TESTS)
 $(ACODEC_TESTS): regtest-aref
 $(VCODEC_TESTS): regtest-vref
 $(LAVF_TESTS) $(LAVFI_TESTS): regtest-ref
 REFFILE = $(SRC_PATH)/tests/ref/$(1)/$(2:regtest-%=%)
 RESFILE = tests/data/$(2:regtest-%=%).$(1).regression
 define CODECTEST_CMD
 	$(SRC_PATH)/tests/codec-regression.sh $@ vsynth1 tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
 	$(SRC_PATH)/tests/codec-regression.sh $@ vsynth2 tests/vsynth2 "$(TARGET_EXEC)" "$(TARGET_PATH)"
 endef
 regtest-ref: regtest-aref regtest-vref
 regtest-vref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm
 	$(CODECTEST_CMD)
 regtest-aref: ffmpeg$(EXESUF) tests/data/asynth1.sw
 	@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
 $(VCODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
 	@echo "TEST VCODEC $(@:regtest-%=%)"
 	@$(CODECTEST_CMD)
 	@diff -u -w $(call REFFILE,vsynth1,$@) $(call RESFILE,vsynth1,$@)
 	@diff -u -w $(call REFFILE,vsynth2,$@) $(call RESFILE,vsynth2,$@)
 $(ACODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
 	@echo "TEST ACODEC $(@:regtest-%=%)"
 	@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
 	@diff -u -w $(call REFFILE,acodec,$@) $(call RESFILE,acodec,$@)
 $(LAVF_TESTS):
 	@echo "TEST LAVF  $(@:regtest-%=%)"
 	@$(SRC_PATH)/tests/lavf-regression.sh $@ lavf tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
 	@diff -u -w $(call REFFILE,lavf,$@) $(call RESFILE,lavf,$@)
 $(LAVFI_TESTS):
 	@echo "TEST LAVFI $(@:regtest-%=%)"
 	@$(SRC_PATH)/tests/lavfi-regression.sh $@ lavfi tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
 	@diff -u -w $(call REFFILE,lavfi,$@) $(call RESFILE,lavfi,$@)
 seektest: codectest lavftest tests/seek_test$(EXESUF)
 	$(SRC_PATH)/tests/seek-regression.sh $(SRC_PATH) "$(TARGET_EXEC)" "$(TARGET_PATH)"
 ffservertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/data/asynth1.sw
 	@echo
 	@echo "Unfortunately ffserver is broken and therefore its regression"
 	@echo "test fails randomly. Treat the results accordingly."
 	@echo
 	$(SRC_PATH)/tests/ffserver-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/ffserver.conf
 tests/vsynth1/00.pgm: tests/videogen$(HOSTEXESUF)
 	mkdir -p tests/vsynth1
 	$(BUILD_ROOT)/$< 'tests/vsynth1/'
 tests/vsynth2/00.pgm: tests/rotozoom$(HOSTEXESUF)
 	mkdir -p tests/vsynth2
 	$(BUILD_ROOT)/$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
 tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF)
 	mkdir -p tests/data
 	$(BUILD_ROOT)/$< $@
 tests/seek_test$(EXESUF): tests/seek_test.o $(FF_DEP_LIBS)
 	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
 ifdef SAMPLES
 include $(SRC_PATH_BARE)/tests/fate.mak
 fate: $(FATE_TESTS)
 $(FATE_TESTS): ffmpeg$(EXESUF)
 	@echo "TEST FATE   $(@:fate-%=%)"
 	@$(SRC_PATH)/tests/fate-run.sh $@ "$(SAMPLES)" "$(TARGET_EXEC)" "$(TARGET_PATH)" '$(CMD)'
 else
 fate:
 	@echo "SAMPLES not specified, cannot run FATE"
 endif
 .PHONY: documentation *test regtest-* zlib-error alltools check config
--- a/12
+++ b/12
@@ -0,0 +1,12 @@
 FFmpeg README
 -------------
 1) Documentation
 ----------------
 * Read the documentation in the doc/ directory.
 2) Licensing
 ------------
 * See the LICENSE file.
--- a/README.md
+++ b/README.md
@@ -1,42 +0,0 @@
 FFmpeg README
 =============
 FFmpeg is a collection of libraries and tools to process multimedia content
 such as audio, video, subtitles and related metadata.
 ## Libraries
 * `libavcodec` provides implementation of a wider range of codecs.
 * `libavformat` implements streaming protocols, container formats and basic I/O access.
 * `libavutil` includes hashers, decompressors and miscellaneous utility functions.
 * `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
 * `libavdevice` provides an abstraction to access capture and playback devices.
 * `libswresample` implements audio mixing and resampling routines.
 * `libswscale` implements color conversion and scaling routines.
 ## Tools
 * [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
  manipulate, convert and stream multimedia content.
 * [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
 * [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
  multimedia content.
 * [ffserver](http://ffmpeg.org/ffserver.html) is a multimedia streaming server
  for live broadcasts.
 * Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
 ## Documentation
 The offline documentation is available in the **doc/** directory.
 The online documentation is available in the main [website](http://ffmpeg.org)
 and in the [wiki](http://trac.ffmpeg.org).
 ### Examples
 Coding examples are available in the **doc/examples** directory.
 ## License
 FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
 GPL. Please refer to the LICENSE file for detailed information.
--- a/179
+++ b/179
@@ -1 +1,178 @@
-2.7.5
+Release Notes
 =============
 * 0.6 "Works with HTML5" June, 2010
 General notes
 -------------
 This release focuses on improvements for the new multimedia elements in HTML5.
 The H.264 and Theora decoders are now significantly faster, the Vorbis decoder
 has seen important updates and this release supports Google's newly released
 libvpx library for the VP8 codec and WebM container.
 Other important changes are additions of decoders including, but not limited to,
 Intel Indeo 5, WMA Pro, WMA Voice and HE-AAC.
 See the Changelog file for a list of significant changes.
 Please note that our policy on bug reports has not changed. We still only accept
 bug reports against HEAD of the FFmpeg trunk repository. If you are experiencing
 any issues with any formally released version of FFmpeg, please try a current
 version of the development code to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
 API and other notable Changes
 -----------------------------
 Please see the file doc/APIchanges for programmer-centric information.
 Notable changes:
 - deprecated vhook subsystem removed
 - deprecated old scaler removed
 - nonfree libamr support for AMR-NB/WB decoding/encoding removed
 - RTMP support in libavformat
 - -formats option split into -formats, -codecs, -bsfs, and -protocols
 - ffprobe tool
 - RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
 - CODEC_CAP_EXPERIMENTAL added
 Added Codecs:
 -------------
 - VQF demuxer
 - PCX encoder
 - CorePNG decoding support
 - 8088flex TMV demuxer and decoder
 - enable symbol versioning by default for linkers that support it
 - V210 decoder and encoder
 - QCP demuxer
 - SoX native format muxer and demuxer
 - AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
 - DPX image decoder
 - Electronic Arts Madcow decoder
 - DivX (XSUB) subtitle encoder
 - experimental AAC encoder
 - Wave64 demuxer
 - IEC-61937 compatible Muxer
 - TwinVQ decoder
 - Bluray (PGS) subtitle decoder
 - LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
 - WMA Pro decoder
 - Core Audio Format demuxer
 - Atrac1 decoder
 - MD STUDIO audio demuxer
 - RF64 support in WAV demuxer
 - MPEG-4 Audio Lossless Coding (ALS) decoder
 - IV8 demuxer
 - CDG demuxer and decoder
 - R210 decoder
 - Auravision Aura 1 and 2 decoders
 - Deluxe Paint Animation playback system
 - SIPR decoder
 - Adobe Filmstrip muxer and demuxer
 - RTP packetization and depacketization of H.263 and AMR
 - Bink demuxer and audio/video decoders
 - IFF PBM/ILBM bitmap decoder
 - Indeo 5 decoder
 - WMA Voice decoder
 - AMR-NB decoder
 - RTSP muxer
 - HE-AAC v1 decoder
 - Kega Game Video (KGV1) decoder
 - Psygnosis YOP demuxer and video decoder
 - RTP hinting in the mov/3gp/mp4 muxer
 - VP8 decoding via libvpx
 Notable license related changes
 -------------------------------
 - remaining GPL parts in AC-3 decoder converted to LGPL
 - libswscale can now be compiled in LGPL mode
 * 0.6.1
 General notes
 -------------
 This point release includes some updates to make the 0.6 release series usable
 for users that need to retain the existing behavior as closely as possible.
 The changes follow below:
 Bugfixes
 --------
 - fix autodetection of E-AC-3 substream samples
 - performance fix for seekable HTTP
 - add missing VP80 fourcc code for the VP8 codec
 - small documentation fixes
 - fix several potentially exploitable issues in the FLIC decoder
  (addresses CVE-2010-3429)
 HE-AAC v2 backport
 ------------------
 This release includes a backport of the AAC decoder from trunk, which
 enables proper playback of HE-AAC v2 media.
 * 0.6.2
 General notes
 -------------
 This is a maintenance-only release that addresses a small number of security
 and portability issues. Distributors and system integrators are encouraged
 to update and share their patches against this branch.
 Security fixes
 --------------
 Programming errors in container and codec implementations may lead to
 denial of service or the execution of arbitrary code if the user is
 tricked into opening a malformed media file or stream.
 Affected and updated have been the implementations of the following
 codecs and container formats:
 - VC1 decoder (Change related to CVE-2011-0723)
 - APE decoder (cf. http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
 * 0.6.3
 General notes
 -------------
 This is a mostly maintenance-only release that addresses a small number
 of bugs such as security and compilation issues. Moreover, this release
 has been updated to work with gcc-4.6 and the VisualOn AAC encoder has
 been backported from the Libav 0.7.2 release. Distributors and system
 integrators are encouraged to update and share their patches against
 this branch.
 For a full list of changes please see the Changelog file.
 * 0.6.4
 General notes
 -------------
 This mostly maintenance-only release that addresses a number a number of
 bugs such as security and compilation issues that have been brought to
 our attention. Among other (rather minor) fixes, this release features
 fixes for the QDM2 decoder (CVE-2011-4351), vp3 decoder (CVE-2011-4352),
 DoS in the VP5/VP6 decoders (CVE-2011-4353), a buffer overflow in the
 Sierra VMD decoder CVE-2011-4364, and a safety fix in the svq1 decoder
 (CVE-2011-4579).
 Distributors and system integrators are encouraged
 to update and share their patches against this branch.  For a full list
 of changes please see the Changelog file.
--- a/15
+++ b/15
@@ -1,15 +0,0 @@
              ┌─────────────────────────────────────┐
              │ RELEASE NOTES for FFmpeg 2.7 "Nash" │
              └─────────────────────────────────────┘
   The FFmpeg Project proudly presents FFmpeg 2.7 "Nash", about 3
   months after the release of FFmpeg 2.6.
   A complete Changelog is available at the root of the project, and the
   complete Git history on http://source.ffmpeg.org.
   We hope you will like this release as much as we enjoyed working on it, and
   as usual, if you have any questions about it, or any FFmpeg related topic,
   feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
   on the mailing-lists.
--- a/1
+++ b/1
@@ -0,0 +1 @@
 0.6.4
--- a/arch.mak
+++ b/arch.mak
@@ -1,17 +0,0 @@
 OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
 OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
 OBJS-$(HAVE_ARMV8)   += $(ARMV8-OBJS)   $(ARMV8-OBJS-yes)
 OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
 OBJS-$(HAVE_LOONGSON3) += $(LOONGSON3-OBJS)  $(LOONGSON3-OBJS-yes)
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)
--- a/cmdutils.c
+++ b/cmdutils.c
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -22,18 +22,11 @@
 #ifndef FFMPEG_CMDUTILS_H
 #define FFMPEG_CMDUTILS_H
-#include <stdint.h>
+#include <inttypes.h>
 #include "config.h"
 #include "libavcodec/avcodec.h"
 #include "libavfilter/avfilter.h"
 #include "libavformat/avformat.h"
 #include "libswscale/swscale.h"
 #ifdef _WIN32
 #undef main /* We don't want SDL to override our main() */
 #endif
 /**
 * program name, defined by the program for show_version().
 */
@@ -44,118 +37,61 @@ extern const char program_name[];
 */
 extern const int program_birth_year;
 extern const int this_year;
 extern const char **opt_names;
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
 extern 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) av_noreturn;
 /**
 * Initialize the cmdutils option system, in particular
 * allocate the *_opts contexts.
 */
 void init_opts(void);
 /**
 * Uninitialize the cmdutils option system, in particular
 * free the *_opts contexts and their contents.
 */
 void uninit_opts(void);
 /**
 * Trivial log callback.
 * Only suitable for opt_help and similar since it lacks prefix handling.
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
 /**
 * 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.
 */
-int opt_default(void *optctx, const char *opt, const char *arg);
+int opt_default(const char *opt, const char *arg);
 /**
- * Set the libav* libraries log level.
+ * Sets the libav* libraries log level.
 */
-int opt_loglevel(void *optctx, const char *opt, const char *arg);
+int opt_loglevel(const char *opt, const char *arg);
 int opt_report(const char *opt);
 int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 int opt_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.
 */
-int opt_timelimit(void *optctx, const char *opt, const char *arg);
+int opt_timelimit(const char *opt, const char *arg);
 /**
- * Parse a string and return its corresponding value as a double.
+ * Parses a string and returns its corresponding value as a double.
- * Exit from the application if the string cannot be correctly
+ * Exits from the application if the string cannot be correctly
 * parsed or the corresponding value is invalid.
 *
 * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
 * @param numstr the string to be parsed
 * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
 * string should be parsed
 * @param min the minimum valid accepted value
 * @param max the maximum valid accepted value
 */
-double parse_number_or_die(const char *context, const char *numstr, int type,
+double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max);
                           double min, double max);
 /**
- * Parse a string specifying a time and return its corresponding
+ * Parses a string specifying a time and returns its corresponding
- * value as a number of microseconds. Exit from the application if
+ * value as a number of microseconds. Exits from the application if
 * the string cannot be correctly parsed.
 *
 * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
 * @param timestr the string to be parsed
 * @param is_duration a flag which tells how to interpret timestr, if
 * not zero timestr is interpreted as a duration, otherwise as a
 * date
 *
- * @see av_parse_time()
+ * @see parse_date()
 */
-int64_t parse_time_or_die(const char *context, const char *timestr,
+int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);
                          int is_duration);
-typedef struct SpecifierOpt {
+typedef struct {
    char *specifier;    /**< stream/chapter/program/... specifier */
    union {
        uint8_t *str;
        int        i;
        int64_t  i64;
        float      f;
        double   dbl;
    } u;
 } SpecifierOpt;
 typedef struct OptionDef {
    const char *name;
    int flags;
 #define HAS_ARG    0x0001
@@ -164,233 +100,42 @@ typedef struct OptionDef {
 #define OPT_STRING 0x0008
 #define OPT_VIDEO  0x0010
 #define OPT_AUDIO  0x0020
 #define OPT_GRAB   0x0040
 #define OPT_INT    0x0080
 #define OPT_FLOAT  0x0100
 #define OPT_SUBTITLE 0x0200
-#define OPT_INT64  0x0400
+#define OPT_FUNC2  0x0400
-#define OPT_EXIT   0x0800
+#define OPT_INT64  0x0800
-#define OPT_DATA   0x1000
+#define OPT_EXIT   0x1000
 #define OPT_PERFILE  0x2000     /* the option is per-file (currently ffmpeg-only).
                                   implied by OPT_OFFSET or OPT_SPEC */
 #define OPT_OFFSET 0x4000       /* option is specified as an offset in a passed optctx */
 #define OPT_SPEC   0x8000       /* option is to be stored in an array of SpecifierOpt.
                                   Implies OPT_OFFSET. Next element after the offset is
                                   an int containing element count in the array. */
 #define OPT_TIME  0x10000
 #define OPT_DOUBLE 0x20000
 #define OPT_INPUT  0x40000
 #define OPT_OUTPUT 0x80000
     union {
-        void *dst_ptr;
+        void (*func_arg)(const char *); //FIXME passing error code as int return would be nicer then exit() in the func
-        int (*func_arg)(void *, const char *, const char *);
+        int *int_arg;
-        size_t off;
+        char **str_arg;
        float *float_arg;
        int (*func2_arg)(const char *, const char *);
        int64_t *int64_arg;
    } u;
    const char *help;
    const char *argname;
 } OptionDef;
-/**
+void show_help_options(const OptionDef *options, const char *msg, int mask, int value);
 * Print help for all options matching specified flags.
 *
 * @param options a list of options
 * @param msg title of this group. Only printed if at least one option matches.
 * @param req_flags print only options which have all those flags set.
 * @param rej_flags don't print options which have any of those flags set.
 * @param alt_flags print only options that have at least one of those flags set
 */
 void show_help_options(const OptionDef *options, const char *msg, int req_flags,
                       int rej_flags, int alt_flags);
 /**
- * Show help for all options with given flags in class and all its
+ * Parses the command line arguments.
 * children.
 */
 void show_help_children(const AVClass *class, int flags);
 /**
 * Per-fftool specific help handler. Implemented in each
 * fftool, called by show_help().
 */
 void show_help_default(const char *opt, const char *arg);
 /**
 * Generic -h handler common to all fftools.
 */
 int show_help(void *optctx, const char *opt, const char *arg);
 /**
 * Parse the command line arguments.
 *
 * @param optctx an opaque options context
 * @param argc   number of command line arguments
 * @param argv   values of command line arguments
 * @param options Array with the definitions required to interpret every
- * option of the form: -option_name [argument]
+ * option of the form: -<option_name> [<argument>]
 * @param parse_arg_function Name of the function called to process every
 * argument without a leading option name flag. NULL if such arguments do
 * not have to be processed.
 */
-void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
+void parse_options(int argc, char **argv, const OptionDef *options,
-                   void (* parse_arg_function)(void *optctx, const char*));
+                   void (* parse_arg_function)(const char*));
 void set_context_opts(void *ctx, void *opts_ctx, int flags);
 /**
- * Parse one given option.
+ * Prints an error message to stderr, indicating filename and a human
 *
 * @return on success 1 if arg was consumed, 0 otherwise; negative number on error
 */
 int parse_option(void *optctx, const char *opt, const char *arg,
                 const OptionDef *options);
 /**
 * An option extracted from the commandline.
 * Cannot use AVDictionary because of options like -map which can be
 * used multiple times.
 */
 typedef struct Option {
    const OptionDef  *opt;
    const char       *key;
    const char       *val;
 } Option;
 typedef struct OptionGroupDef {
    /**< group name */
    const char *name;
    /**
     * Option to be used as group separator. Can be NULL for groups which
     * are terminated by a non-option argument (e.g. ffmpeg output files)
     */
    const char *sep;
    /**
     * Option flags that must be set on each option that is
     * applied to this group
     */
    int flags;
 } OptionGroupDef;
 typedef struct OptionGroup {
    const OptionGroupDef *group_def;
    const char *arg;
    Option *opts;
    int  nb_opts;
    AVDictionary *codec_opts;
    AVDictionary *format_opts;
    AVDictionary *resample_opts;
    struct SwsContext *sws_opts;
    AVDictionary *swr_opts;
 } OptionGroup;
 /**
 * A list of option groups that all have the same group type
 * (e.g. input files or output files)
 */
 typedef struct OptionGroupList {
    const OptionGroupDef *group_def;
    OptionGroup *groups;
    int       nb_groups;
 } OptionGroupList;
 typedef struct OptionParseContext {
    OptionGroup global_opts;
    OptionGroupList *groups;
    int           nb_groups;
    /* parsing state */
    OptionGroup cur_group;
 } OptionParseContext;
 /**
 * Parse an options group and write results into optctx.
 *
 * @param optctx an app-specific options context. NULL for global options group
 */
 int parse_optgroup(void *optctx, OptionGroup *g);
 /**
 * Split the commandline into an intermediate form convenient for further
 * processing.
 *
 * The commandline is assumed to be composed of options which either belong to a
 * group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
 * (everything else).
 *
 * A group (defined by an OptionGroupDef struct) is a sequence of options
 * terminated by either a group separator option (e.g. -i) or a parameter that
 * is not an option (doesn't start with -). A group without a separator option
 * must always be first in the supplied groups list.
 *
 * All options within the same group are stored in one OptionGroup struct in an
 * OptionGroupList, all groups with the same group definition are stored in one
 * OptionGroupList in OptionParseContext.groups. The order of group lists is the
 * same as the order of group definitions.
 */
 int split_commandline(OptionParseContext *octx, int argc, char *argv[],
                      const OptionDef *options,
                      const OptionGroupDef *groups, int nb_groups);
 /**
 * Free all allocated memory in an OptionParseContext.
 */
 void uninit_parse_context(OptionParseContext *octx);
 /**
 * Find the '-loglevel' option in the command line args and apply it.
 */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);
 /**
 * Return index of option opt in argv or 0 if not found.
 */
 int locate_option(int argc, char **argv, const OptionDef *options,
                  const char *optname);
 /**
 * Check if the given stream matches a stream specifier.
 *
 * @param s  Corresponding format context.
 * @param st Stream from s to be checked.
 * @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
 *
 * @return 1 if the stream matches, 0 if it doesn't, <0 on error
 */
 int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
 /**
 * Filter out options for given codec.
 *
 * Create a new options dictionary containing only the options from
 * opts which apply to the codec with ID codec_id.
 *
 * @param opts     dictionary to place options in
 * @param codec_id ID of the codec that should be filtered for
 * @param s Corresponding format context.
 * @param st A stream from s for which the options should be filtered.
 * @param codec The particular codec for which the options should be filtered.
 *              If null, the default one is looked up according to the codec id.
 * @return a pointer to the created dictionary
 */
 AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
                                AVFormatContext *s, AVStream *st, AVCodec *codec);
 /**
 * Setup AVCodecContext options for avformat_find_stream_info().
 *
 * Create an array of dictionaries, one dictionary for each stream
 * contained in s.
 * Each dictionary will contain the options from codec_opts which can
 * be applied to the corresponding stream codec context.
 *
 * @return pointer to the created array of dictionaries, NULL if it
 * cannot be created
 */
 AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
                                           AVDictionary *codec_opts);
 /**
 * Print an error message to stderr, indicating filename and a human
 * readable description of the error code err.
 *
 * If strerror_r() is not available the use of this function in a
@@ -400,203 +145,79 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
 */
 void print_error(const char *filename, int err);
 void list_fmts(void (*get_fmt_string)(char *buf, int buf_size, int fmt), int nb_fmts);
 /**
- * Print the program banner to stderr. The banner contents depend on the
+ * Prints the program banner to stderr. The banner contents depend on the
 * current version of the repository and of the libav* libraries used by
 * the program.
 */
-void show_banner(int argc, char **argv, const OptionDef *options);
+void show_banner(void);
 /**
- * Print the version of the program to stdout. The version message
+ * Prints the version of the program to stdout. The version message
 * depends on the current versions of the repository and of the libav*
 * libraries.
 * This option processing function does not utilize the arguments.
 */
-int show_version(void *optctx, const char *opt, const char *arg);
+void show_version(void);
 /**
- * Print the build configuration of the program to stdout. The contents
+ * Prints the license of the program to stdout. The license depends on
 * 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.
 * This option processing function does not utilize the arguments.
 */
-int show_license(void *optctx, const char *opt, const char *arg);
+void show_license(void);
 /**
- * Print a listing containing all the formats supported by the
+ * Prints a listing containing all the formats supported by the
 * program (including devices).
 * This option processing function does not utilize the arguments.
 */
 int show_formats(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the devices supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_devices(void *optctx, const char *opt, const char *arg);
 #if CONFIG_AVDEVICE
 /**
 * Print a listing containing audodetected sinks of the output device.
 * Device name with options may be passed as an argument to limit results.
 */
 int show_sinks(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing audodetected sources of the input device.
 * Device name with options may be passed as an argument to limit results.
 */
 int show_sources(void *optctx, const char *opt, const char *arg);
 #endif
 /**
 * Print a listing containing all the codecs supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_codecs(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the decoders supported by the
 * program.
 */
-int show_decoders(void *optctx, const char *opt, const char *arg);
+void show_formats(void);
 /**
- * Print a listing containing all the encoders supported by the
+ * Prints a listing containing all the codecs supported by the
 * program.
 */
-int show_encoders(void *optctx, const char *opt, const char *arg);
+void show_codecs(void);
 /**
- * Print a listing containing all the filters supported by the
+ * Prints a listing containing all the filters supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_filters(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the bit stream filters supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_bsfs(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the protocols supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_protocols(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the pixel formats supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_pix_fmts(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the standard channel layouts supported by
 * the program.
 * This option processing function does not utilize the arguments.
 */
 int show_layouts(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the sample formats supported by the
 * program.
 */
-int show_sample_fmts(void *optctx, const char *opt, const char *arg);
+void show_filters(void);
 /**
- * Print a listing containing all the color names and values recognized
+ * Prints a listing containing all the bit stream filters supported by the
- * by the program.
+ * program.
 */
-int show_colors(void *optctx, const char *opt, const char *arg);
+void show_bsfs(void);
 /**
- * Return a positive value if a line read from standard input
+ * Prints a listing containing all the protocols supported by the
- * starts with [yY], otherwise return 0.
+ * program.
 */
 void show_protocols(void);
 /**
 * Prints a listing containing all the pixel formats supported by the
 * program.
 */
 void show_pix_fmts(void);
 /**
 * Returns a positive value if reads from standard input a line
 * starting with [yY], otherwise returns 0.
 */
 int read_yesno(void);
 /**
- * Read the file with name filename, and put its content in a newly
+ * Reads the file with name filename, and puts its content in a newly
 * allocated 0-terminated buffer.
 *
- * @param filename file to read from
+ * @param bufptr puts here the pointer to the newly allocated buffer
- * @param bufptr location where pointer to buffer is returned
+ * @param size puts here the size of the newly allocated buffer
- * @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);
+int read_file(const char *filename, char **bufptr, size_t *size);
-/**
+#endif /* FFMPEG_CMDUTILS_H */
 * Get a file corresponding to a preset file.
 *
 * If is_path is non-zero, look for the file in the path preset_name.
 * Otherwise search for a file named arg.ffpreset in the directories
 * $FFMPEG_DATADIR (if set), $HOME/.ffmpeg, and in the datadir defined
 * at configuration time or in a "ffpresets" folder along the executable
 * on win32, in that order. If no such file is found and
 * codec_name is defined, then search for a file named
 * codec_name-preset_name.avpreset in the above-mentioned directories.
 *
 * @param filename buffer where the name of the found filename is written
 * @param filename_size size in bytes of the filename buffer
 * @param preset_name name of the preset to search
 * @param is_path tell if preset_name is a filename path
 * @param codec_name name of the codec for which to look for the
 * preset, may be NULL
 */
 FILE *get_preset_file(char *filename, size_t filename_size,
                      const char *preset_name, int is_path, const char *codec_name);
 /**
 * Realloc array to hold new_size elements of elem_size.
 * Calls exit() on failure.
 *
 * @param array array to reallocate
 * @param elem_size size in bytes of each element
 * @param size new element count will be written here
 * @param new_size number of elements to place in reallocated array
 * @return reallocated array
 */
 void *grow_array(void *array, int elem_size, int *size, int new_size);
 #define 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)
 #define GET_PIX_FMT_NAME(pix_fmt)\
    const char *name = av_get_pix_fmt_name(pix_fmt);
 #define GET_SAMPLE_FMT_NAME(sample_fmt)\
    const char *name = av_get_sample_fmt_name(sample_fmt)
 #define GET_SAMPLE_RATE_NAME(rate)\
    char name[16];\
    snprintf(name, sizeof(name), "%d", rate);
 #define GET_CH_LAYOUT_NAME(ch_layout)\
    char name[16];\
    snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
 #define GET_CH_LAYOUT_DESC(ch_layout)\
    char name[128];\
    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
 double get_rotation(AVStream *st);
 #endif /* CMDUTILS_H */
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -1,35 +1,13 @@
-    { "L"          , OPT_EXIT, {.func_arg = show_license},      "show license" },
+    { "L", OPT_EXIT, {(void*)show_license}, "show license" },
-    { "h"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "h", OPT_EXIT, {(void*)show_help}, "show help" },
-    { "?"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "?", OPT_EXIT, {(void*)show_help}, "show help" },
-    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "help", OPT_EXIT, {(void*)show_help}, "show help" },
-    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "-help", OPT_EXIT, {(void*)show_help}, "show help" },
-    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
+    { "version", OPT_EXIT, {(void*)show_version}, "show version" },
-    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
+    { "formats"  , OPT_EXIT, {(void*)show_formats  }, "show available formats" },
-    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
+    { "codecs"   , OPT_EXIT, {(void*)show_codecs   }, "show available codecs" },
-    { "devices"    , OPT_EXIT, {.func_arg = show_devices  },    "show available devices" },
+    { "bsfs"     , OPT_EXIT, {(void*)show_bsfs     }, "show available bit stream filters" },
-    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
+    { "protocols", OPT_EXIT, {(void*)show_protocols}, "show available protocols" },
-    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
+    { "filters",   OPT_EXIT, {(void*)show_filters  }, "show available filters" },
-    { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
+    { "pix_fmts" , OPT_EXIT, {(void*)show_pix_fmts }, "show available pixel formats" },
-    { "bsfs"       , OPT_EXIT, {.func_arg = show_bsfs     },    "show available bit stream filters" },
+    { "loglevel", HAS_ARG | OPT_FUNC2, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
    { "protocols"  , OPT_EXIT, {.func_arg = show_protocols},    "show available protocols" },
    { "filters"    , OPT_EXIT, {.func_arg = show_filters  },    "show available filters" },
    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
    { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
    { "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" },
    { "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
 #if CONFIG_AVDEVICE
    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
      "list sources of the input device", "device" },
    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
      "list sinks of the output device", "device" },
 #endif
--- a/cmdutils_opencl.c
+++ b/cmdutils_opencl.c
@@ -1,276 +0,0 @@
 /*
 * 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 "libavutil/avstring.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_relative();
    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_relative() - 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;
                av_strlcpy(devices[count].device_name, device_node->device_name,
                           sizeof(devices[count].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;
 }
--- a/common.mak
+++ b/common.mak
@@ -5,89 +5,61 @@
 # first so "all" becomes default target
 all: all-yes
 DEFAULT_YASMD=.dbg
 ifeq ($(DBG),1)
 YASMD=$(DEFAULT_YASMD)
 else
 YASMD=
 endif
 ifndef SUBDIR
 vpath %.c   $(SRC_DIR)
 vpath %.h   $(SRC_DIR)
 vpath %.S   $(SRC_DIR)
 vpath %.asm $(SRC_DIR)
 vpath %.v   $(SRC_DIR)
 ifeq ($(SRC_DIR),$(SRC_PATH_BARE))
 BUILD_ROOT_REL = .
 else
 BUILD_ROOT_REL = ..
 endif
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
+BRIEF  = CC AS YASM AR LD HOSTCC STRIP CP
-SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
+SILENT = DEPCC YASMDEP RM RANLIB
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
-    $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
+    $(eval $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
-$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
+$(foreach VAR,$(SILENT),$(eval $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
-# NASM requires -I path terminated with /
+CPPFLAGS := -I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(CPPFLAGS)
-IFLAGS     := -I. -I$(SRC_PATH)/
+CFLAGS   += $(ECFLAGS)
 CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
 CFLAGS     += $(ECFLAGS)
 CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
 YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
 define COMPILE
       $(call $(1)DEP,$(1))
       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
 endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
 COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.c
-	$(COMPILE_C)
+	$(CCDEP)
-
+	$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_DEPFLAGS) -c $(CC_O) $<
 %.o: %.cpp
 	$(COMPILE_CXX)
 %.o: %.m
 	$(COMPILE_C)
 %.s: %.c
 	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
 %.o: %.S
-	$(COMPILE_S)
+	$(ASDEP)
 	$(AS) $(CPPFLAGS) $(ASFLAGS) $(AS_DEPFLAGS) -c -o $@ $<
-%_host.o: %.c
+%.ho: %.h
-	$(COMPILE_HOSTC)
+	$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
-%.o: %.rc
+%$(EXESUF): %.c
 	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@
 %.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
 %.c %.h: TAG = GEN
-# Dummy rule to stop make trying to rebuild removed or renamed headers
+install: install-libs install-headers
-%.h:
+install-libs: install-libs-yes
-	@:
+
 uninstall: uninstall-libs uninstall-headers
 .PHONY: all depend dep *clean install* uninstall* examples testprogs
 # Disable suffix rules.  Most of the builtin rules are suffix rules,
 # so this saves some time on slow systems.
@@ -97,66 +69,41 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 $(OBJS):
 endif
-include $(SRC_PATH)/arch.mak
+OBJS-$(HAVE_MMX) +=  $(MMX-OBJS-yes)
 CFLAGS    += $(CFLAGS-yes)
 OBJS      += $(OBJS-yes)
-SLIBOBJS  += $(SLIBOBJS-yes)
+FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
-LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
+FFEXTRALIBS := $(addprefix -l,$(addsuffix $(BUILDSUF),$(FFLIBS))) $(EXTRALIBS)
-FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
+FFLDFLAGS   := $(addprefix -L$(BUILD_ROOT)/lib,$(ALLFFLIBS)) $(LDFLAGS)
-OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
+EXAMPLES  := $(addprefix $(SUBDIR),$(addsuffix -example$(EXESUF),$(EXAMPLES)))
-SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
+OBJS      := $(addprefix $(SUBDIR),$(sort $(OBJS)))
-TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
+TESTOBJS  := $(addprefix $(SUBDIR),$(TESTOBJS))
-TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
+TESTPROGS := $(addprefix $(SUBDIR),$(addsuffix -test$(EXESUF),$(TESTPROGS)))
-HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
+HOSTOBJS  := $(addprefix $(SUBDIR),$(addsuffix .o,$(HOSTPROGS)))
-HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
+HOSTPROGS := $(addprefix $(SUBDIR),$(addsuffix $(HOSTEXESUF),$(HOSTPROGS)))
 TOOLS     += $(TOOLS-yes)
 TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)
-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
+DEP_LIBS := $(foreach NAME,$(FFLIBS),$(BUILD_ROOT_REL)/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 += $(addprefix $(ARCH)/,$(ARCH_HEADERS))
-SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
+SKIPHEADERS := $(addprefix $(SUBDIR),$(SKIPHEADERS-) $(SKIPHEADERS))
-HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
+checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
 checkheaders: $(HOBJS)
 .SECONDARY:   $(HOBJS:.o=.c)
 alltools: $(TOOLS)
 $(HOSTOBJS): %.o: %.c
-	$(COMPILE_HOSTC)
+	$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
+	$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
-$(OBJS):     | $(sort $(dir $(OBJS)))
+DEPS := $(OBJS:.o=.d)
-$(HOBJS):    | $(sort $(dir $(HOBJS)))
+depend dep: $(DEPS)
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
+CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver
 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda *$(DEFAULT_YASMD).asm
 DISTCLEANSUFFIXES = *.pc
-LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
+LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
-define RULES
+-include $(wildcard $(DEPS))
 clean::
 	$(RM) $(OBJS) $(OBJS:.o=.d) $(OBJS:.o=$(DEFAULT_YASMD).d)
 	$(RM) $(HOSTPROGS)
 	$(RM) $(TOOLS)
 endef
 $(eval $(RULES))
 -include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -1,31 +0,0 @@
 /*
 * 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 */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -1,882 +0,0 @@
 // 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__
 #include "avs/config.h"
 #include "avs/capi.h"
 #include "avs/types.h"
 /////////////////////////////////////////////////////////////////////
 //
 // Constants
 //
 #ifndef __AVISYNTH_6_H__
 enum { AVISYNTH_INTERFACE_VERSION = 6 };
 #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  = 8,
      AVS_CS_SHIFT_SAMPLE_BITS = 16,
      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 {
  // New 2.6 explicitly defined cache hints.
  AVS_CACHE_NOTHING=10, // Do not cache video.
  AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
  AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
  AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
  AVS_CACHE_GET_POLICY=30, // Get the current policy.
  AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
  AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
  AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
  AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
  AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
  AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
  AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
  AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
  AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
  AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
  AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
  AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
  AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
  AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
  AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
  AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
  AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
  AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
  AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
  AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
  AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
    AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
    AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
    AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
    AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
    AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
  AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
    AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
    AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
    AVS_CACHE_THREAD_SAFE=243, //  Allow all threads in any instance.
    AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
  AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
    AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
    AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
    AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
  };
 #ifdef BUILDING_AVSCORE
 struct AVS_ScriptEnvironment {
        IScriptEnvironment * env;
        const char * error;
        AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
                : env(e), error(0) {}
 };
 #endif
 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_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
 AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
 AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
 AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
 AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
 AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
        { return ((p->image_type & property)==property ); }
 AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
        { return !!(p->pixel_type & AVS_CS_PLANAR); }
 AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int 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_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
 AVSC_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
 AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
 AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
 AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
 AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
 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;
 }
 #ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 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));
 }
 #endif
 /////////////////////////////////////////////////////////////////////
 //
 // 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_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
 #ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
        return avs_get_pitch_p(p, 0);}
 #endif
 AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
 AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
        return p->row_size; }
 AVSC_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
 AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
        return p->height;}
 AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
 #ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
        return avs_get_read_ptr_p(p, 0);}
 #endif
 AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
 AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
 #ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
        return avs_get_write_ptr_p(p, 0);}
 #endif
 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_AVX        = 0x800,   //  Sandy Bridge, Bulldozer
  AVS_CPUF_SSE4_2    = 0x1000,   //  Nehalem
 //AVS_CPUF_AVX2      = 0x2000,   //  Haswell
 //AVS_CPUF_AVX512    = 0x4000,   //  Knights Landing
 };
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
 AVSC_API(int, 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,FRAME_ALIGN);}
 AVSC_INLINE
 AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
                               const AVS_VideoInfo * vi)
  {return avs_new_video_frame_a(env,vi,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_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);
  AVSC_DECLARE_FUNC(avs_get_error);
  AVSC_DECLARE_FUNC(avs_is_yv24);
  AVSC_DECLARE_FUNC(avs_is_yv16);
  AVSC_DECLARE_FUNC(avs_is_yv12);
  AVSC_DECLARE_FUNC(avs_is_yv411);
  AVSC_DECLARE_FUNC(avs_is_y8);
  AVSC_DECLARE_FUNC(avs_is_color_space);
  AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
  AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
  AVSC_DECLARE_FUNC(avs_bits_per_pixel);
  AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
  AVSC_DECLARE_FUNC(avs_row_size);
  AVSC_DECLARE_FUNC(avs_bmp_size);
  AVSC_DECLARE_FUNC(avs_get_pitch_p);
  AVSC_DECLARE_FUNC(avs_get_row_size_p);
  AVSC_DECLARE_FUNC(avs_get_height_p);
  AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
  AVSC_DECLARE_FUNC(avs_is_writable);
  AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
 };
 #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_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);
  AVSC_LOAD_FUNC(avs_get_error);
  AVSC_LOAD_FUNC(avs_is_yv24);
  AVSC_LOAD_FUNC(avs_is_yv16);
  AVSC_LOAD_FUNC(avs_is_yv12);
  AVSC_LOAD_FUNC(avs_is_yv411);
  AVSC_LOAD_FUNC(avs_is_y8);
  AVSC_LOAD_FUNC(avs_is_color_space);
  AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
  AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
  AVSC_LOAD_FUNC(avs_bits_per_pixel);
  AVSC_LOAD_FUNC(avs_bytes_from_pixels);
  AVSC_LOAD_FUNC(avs_row_size);
  AVSC_LOAD_FUNC(avs_bmp_size);
  AVSC_LOAD_FUNC(avs_get_pitch_p);
  AVSC_LOAD_FUNC(avs_get_row_size_p);
  AVSC_LOAD_FUNC(avs_get_height_p);
  AVSC_LOAD_FUNC(avs_get_read_ptr_p);
  AVSC_LOAD_FUNC(avs_is_writable);
  AVSC_LOAD_FUNC(avs_get_write_ptr_p);
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC
  return library;
 fail:
  free(library);
  return NULL;
 }
 AVSC_INLINE void avs_free_library(AVS_Library *library) {
  if (library == NULL)
    return;
  FreeLibrary(library->handle);
  free(library);
 }
 #endif
 #endif
--- a/compat/avisynth/avs/capi.h
+++ b/compat/avisynth/avs/capi.h
@@ -1,62 +0,0 @@
 // Avisynth C Interface Version 0.20
 // Copyright 2003 Kevin Atkinson
 // This program is free software; you can redistribute it and/or modify
 // it under the terms of the GNU General Public License as published by
 // the Free Software Foundation; either version 2 of the License, or
 // (at your option) any later version.
 //
 // This program is distributed in the hope that it will be useful,
 // but WITHOUT ANY WARRANTY; without even the implied warranty of
 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 // GNU General Public License for more details.
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
 // Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
 // Avisynth C interface with independent modules that communicate with
 // the Avisynth C interface solely through the interfaces defined in
 // avisynth_c.h, regardless of the license terms of these independent
 // modules, and to copy and distribute the resulting combined work
 // under terms of your choice, provided that every copy of the
 // combined work is accompanied by a complete copy of the source code
 // of the Avisynth C interface and Avisynth itself (with the version
 // used to produce the combined work), being distributed under the
 // terms of the GNU General Public License plus this exception.  An
 // independent module is a module which is not derived from or based
 // on Avisynth C Interface, such as 3rd-party filters, import and
 // export plugins, or graphical user interfaces.
 #ifndef AVS_CAPI_H
 #define AVS_CAPI_H
 #ifdef __cplusplus
 #  define EXTERN_C extern "C"
 #else
 #  define EXTERN_C
 #endif
 #ifndef AVSC_USE_STDCALL
 #  define AVSC_CC __cdecl
 #else
 #  define AVSC_CC __stdcall
 #endif
 #define AVSC_INLINE static __inline
 #ifdef BUILDING_AVSCORE
 #  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
 #endif //AVS_CAPI_H
--- a/compat/avisynth/avs/config.h
+++ b/compat/avisynth/avs/config.h
@@ -1,55 +0,0 @@
 // Avisynth C Interface Version 0.20
 // Copyright 2003 Kevin Atkinson
 // This program is free software; you can redistribute it and/or modify
 // it under the terms of the GNU General Public License as published by
 // the Free Software Foundation; either version 2 of the License, or
 // (at your option) any later version.
 //
 // This program is distributed in the hope that it will be useful,
 // but WITHOUT ANY WARRANTY; without even the implied warranty of
 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 // GNU General Public License for more details.
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
 // Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
 // Avisynth C interface with independent modules that communicate with
 // the Avisynth C interface solely through the interfaces defined in
 // avisynth_c.h, regardless of the license terms of these independent
 // modules, and to copy and distribute the resulting combined work
 // under terms of your choice, provided that every copy of the
 // combined work is accompanied by a complete copy of the source code
 // of the Avisynth C interface and Avisynth itself (with the version
 // used to produce the combined work), being distributed under the
 // terms of the GNU General Public License plus this exception.  An
 // independent module is a module which is not derived from or based
 // on Avisynth C Interface, such as 3rd-party filters, import and
 // export plugins, or graphical user interfaces.
 #ifndef AVS_CONFIG_H
 #define AVS_CONFIG_H
 // Undefine this to get cdecl calling convention
 #define AVSC_USE_STDCALL 1
 // NOTE TO PLUGIN AUTHORS:
 // Because FRAME_ALIGN can be substantially higher than the alignment
 // a plugin actually needs, plugins should not use FRAME_ALIGN to check for
 // alignment. They should always request the exact alignment value they need.
 // This is to make sure that plugins work over the widest range of AviSynth
 // builds possible.
 #define FRAME_ALIGN 32
 #if   defined(_M_AMD64) || defined(__x86_64)
 #   define X86_64
 #elif defined(_M_IX86) || defined(__i386__)
 #   define X86_32
 #else
 #   error Unsupported CPU architecture.
 #endif
 #endif //AVS_CONFIG_H
--- a/compat/avisynth/avs/types.h
+++ b/compat/avisynth/avs/types.h
@@ -1,51 +0,0 @@
 // Avisynth C Interface Version 0.20
 // Copyright 2003 Kevin Atkinson
 // This program is free software; you can redistribute it and/or modify
 // it under the terms of the GNU General Public License as published by
 // the Free Software Foundation; either version 2 of the License, or
 // (at your option) any later version.
 //
 // This program is distributed in the hope that it will be useful,
 // but WITHOUT ANY WARRANTY; without even the implied warranty of
 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 // GNU General Public License for more details.
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
 // Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
 // Avisynth C interface with independent modules that communicate with
 // the Avisynth C interface solely through the interfaces defined in
 // avisynth_c.h, regardless of the license terms of these independent
 // modules, and to copy and distribute the resulting combined work
 // under terms of your choice, provided that every copy of the
 // combined work is accompanied by a complete copy of the source code
 // of the Avisynth C interface and Avisynth itself (with the version
 // used to produce the combined work), being distributed under the
 // terms of the GNU General Public License plus this exception.  An
 // independent module is a module which is not derived from or based
 // on Avisynth C Interface, such as 3rd-party filters, import and
 // export plugins, or graphical user interfaces.
 #ifndef AVS_TYPES_H
 #define AVS_TYPES_H
 // Define all types necessary for interfacing with avisynth.dll
 // Raster types used by VirtualDub & Avisynth
 typedef unsigned int    Pixel32;
 typedef unsigned char   BYTE;
 // Audio Sample information
 typedef float SFLOAT;
 #ifdef __GNUC__
 typedef long long int INT64;
 #else
 typedef __int64 INT64;
 #endif
 #endif //AVS_TYPES_H
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@@ -1,728 +0,0 @@
 // 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 = {0}; 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 = {0}; v.type = 'i'; v.d.integer = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
        { AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_float(float v0)
        { AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
 AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
        { AVS_Value v = {0}; 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 = {0}; 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 = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_Clip
 //
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(void, avs_release_clip)(AVS_Clip *);
 AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
 AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
 AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
 AVSC_API(int, avs_get_version)(AVS_Clip *);
 AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
 // The returned video frame must be released with avs_release_video_frame
 AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
 // return field parity if field_based, else parity of first field in frame
 AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
                                  INT64 start, INT64 count);
 // start and count are in samples
 AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
                                        int cachehints, size_t frame_range);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 // This is the callback type used by avs_add_function
 typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
 typedef struct AVS_FilterInfo AVS_FilterInfo;
 struct AVS_FilterInfo
 {
  // these members should not be modified outside of the AVS_ApplyFunc callback
  AVS_Clip * child;
  AVS_VideoInfo vi;
  AVS_ScriptEnvironment * env;
  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
                                  INT64 start, INT64 count);
  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
                                        int frame_range);
  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
  // Should be set when ever there is an error to report.
  // It is cleared before any of the above methods are called
  const char * error;
  // this is to store whatever and may be modified at will
  void * user_data;
 };
 // Create a new filter
 // fi is set to point to the AVS_FilterInfo so that you can
 //   modify it once it is initilized.
 // store_child should generally be set to true.  If it is not
 //    set than ALL methods (the function pointers) must be defined
 // If it is set than you do not need to worry about freeing the child
 //    clip.
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
                                      AVS_FilterInfo * * fi,
                                      AVS_Value child, int store_child);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_ScriptEnvironment
 //
 // For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
 enum {
                                /* slowest CPU to support extension */
  AVS_CPU_FORCE        = 0x01,   // N/A
  AVS_CPU_FPU          = 0x02,   // 386/486DX
  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
  AVS_CPU_3DNOW        = 0x40,   // K6-2
  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
                                 // which only Hammer will have anyway)
 };
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
 AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
 AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
 AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
 AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
 AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
 // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
 AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
                                     const char * name, const char * params,
                                     AVS_ApplyFunc apply, void * user_data);
 AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
 AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
                               AVS_Value args, const char** arg_names);
 // The returned value must be be released with avs_release_value
 AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
 // The returned value must be be released with avs_release_value
 AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
 AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
 //void avs_push_context(AVS_ScriptEnvironment *, int level=0);
 //void avs_pop_context(AVS_ScriptEnvironment *);
 AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
                                          const AVS_VideoInfo * vi, int align);
 // align should be at least 16
 #if defined __cplusplus
 }
 #endif // __cplusplus
 #ifndef AVSC_NO_DECLSPEC
 AVSC_INLINE
 AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                     const AVS_VideoInfo * vi)
  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 AVSC_INLINE
 AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
                               const AVS_VideoInfo * vi)
  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 #endif
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
 AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
 typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
 AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
 AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
 // The returned video frame must be be released
 AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
 AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
 // avisynth.dll exports this; it's a way to use it as a library, without
 // writing an AVS script or without going through AVIFile.
 AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 // this symbol is the entry point for the plugin and must
 // be defined
 AVSC_EXPORT
 const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
 AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
 // The returned video frame must be be released
 #if defined __cplusplus
 }
 #endif // __cplusplus
 #endif //__AVXSYNTH_C__
--- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h
+++ b/compat/avisynth/windowsPorts/basicDataTypeConversions.h
@@ -1,85 +0,0 @@
 #ifndef __DATA_TYPE_CONVERSIONS_H__
 #define __DATA_TYPE_CONVERSIONS_H__
 #include <stdint.h>
 #include <wchar.h>
 #ifdef __cplusplus
 namespace avxsynth {
 #endif // __cplusplus
 typedef int64_t __int64;
 typedef int32_t __int32;
 #ifdef __cplusplus
 typedef bool BOOL;
 #else
 typedef uint32_t BOOL;
 #endif // __cplusplus
 typedef void* HMODULE;
 typedef void* LPVOID;
 typedef void* PVOID;
 typedef PVOID HANDLE;
 typedef HANDLE HWND;
 typedef HANDLE HINSTANCE;
 typedef void* HDC;
 typedef void* HBITMAP;
 typedef void* HICON;
 typedef void* HFONT;
 typedef void* HGDIOBJ;
 typedef void* HBRUSH;
 typedef void* HMMIO;
 typedef void* HACMSTREAM;
 typedef void* HACMDRIVER;
 typedef void* HIC;
 typedef void* HACMOBJ;
 typedef HACMSTREAM* LPHACMSTREAM;
 typedef void* HACMDRIVERID;
 typedef void* LPHACMDRIVER;
 typedef unsigned char BYTE;
 typedef BYTE* LPBYTE;
 typedef char TCHAR;
 typedef TCHAR* LPTSTR;
 typedef const TCHAR* LPCTSTR;
 typedef char* LPSTR;
 typedef LPSTR LPOLESTR;
 typedef const char* LPCSTR;
 typedef LPCSTR LPCOLESTR;
 typedef wchar_t WCHAR;
 typedef unsigned short WORD;
 typedef unsigned int UINT;
 typedef UINT MMRESULT;
 typedef uint32_t DWORD;
 typedef DWORD COLORREF;
 typedef DWORD FOURCC;
 typedef DWORD HRESULT;
 typedef DWORD* LPDWORD;
 typedef DWORD* DWORD_PTR;
 typedef int32_t LONG;
 typedef int32_t* LONG_PTR;
 typedef LONG_PTR LRESULT;
 typedef uint32_t ULONG;
 typedef uint32_t* ULONG_PTR;
 //typedef __int64_t intptr_t;
 typedef uint64_t _fsize_t;
 //
 // Structures
 //
 typedef struct _GUID {
  DWORD Data1;
  WORD  Data2;
  WORD  Data3;
  BYTE  Data4[8];
 } GUID;
 typedef GUID REFIID;
 typedef GUID CLSID;
 typedef CLSID* LPCLSID;
 typedef GUID IID;
 #ifdef __cplusplus
 }; // namespace avxsynth
 #endif // __cplusplus
 #endif //  __DATA_TYPE_CONVERSIONS_H__
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -1,77 +0,0 @@
 #ifndef __WINDOWS2LINUX_H__
 #define __WINDOWS2LINUX_H__
 /*
 * LINUX SPECIFIC DEFINITIONS
 */
 //
 // Data types conversions
 //
 #include <stdlib.h>
 #include <string.h>
 #include "basicDataTypeConversions.h"
 #ifdef __cplusplus
 namespace avxsynth {
 #endif // __cplusplus
 //
 // purposefully define the following MSFT definitions
 // to mean nothing (as they do not mean anything on Linux)
 //
 #define __stdcall
 #define __cdecl
 #define noreturn
 #define __declspec(x)
 #define STDAPI       extern "C" HRESULT
 #define STDMETHODIMP HRESULT __stdcall
 #define STDMETHODIMP_(x) x __stdcall
 #define STDMETHOD(x)    virtual HRESULT x
 #define STDMETHOD_(a, x) virtual a x
 #ifndef TRUE
 #define TRUE  true
 #endif
 #ifndef FALSE
 #define FALSE false
 #endif
 #define S_OK                (0x00000000)
 #define S_FALSE             (0x00000001)
 #define E_NOINTERFACE       (0X80004002)
 #define E_POINTER           (0x80004003)
 #define E_FAIL              (0x80004005)
 #define E_OUTOFMEMORY       (0x8007000E)
 #define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
 #define FAILED(hr)              ((hr) & 0x80000000)
 #define SUCCEEDED(hr)           (!FAILED(hr))
 //
 // Functions
 //
 #define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
 #define MAKEWORD(a,b) (((a) << 8) | (b))
 #define lstrlen                             strlen
 #define lstrcpy                             strcpy
 #define lstrcmpi                            strcasecmp
 #define _stricmp                            strcasecmp
 #define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
 #define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
 // Windows uses (new, old) ordering but GCC has (old, new)
 #define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
 #define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
 #define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
 #define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
 #define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
 #ifdef __cplusplus
 }; // namespace avxsynth
 #endif // __cplusplus
 #endif //  __WINDOWS2LINUX_H__
--- a/compat/float/float.h
+++ b/compat/float/float.h
@@ -1,35 +0,0 @@
 /*
 * Work around broken floating point limits on some systems.
 *
 * 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_next <float.h>
 #ifdef FLT_MAX
 #undef  FLT_MAX
 #define FLT_MAX 3.40282346638528859812e+38F
 #undef  FLT_MIN
 #define FLT_MIN 1.17549435082228750797e-38F
 #undef  DBL_MAX
 #define DBL_MAX ((double)1.79769313486231570815e+308L)
 #undef  DBL_MIN
 #define DBL_MIN ((double)2.22507385850720138309e-308L)
 #endif
--- a/compat/float/limits.h
+++ b/compat/float/limits.h
@@ -1,22 +0,0 @@
 /*
 * Work around broken floating point limits on some systems.
 *
 * 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_next <limits.h>
 #include <float.h>
--- a/compat/getopt.c
+++ b/compat/getopt.c
@@ -1,84 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 /*
 * This file was copied from the following newsgroup posting:
 *
 * Newsgroups: mod.std.unix
 * Subject: public domain AT&T getopt source
 * Date: 3 Nov 85 19:34:15 GMT
 *
 * Here's something you've all been waiting for:  the AT&T public domain
 * source for getopt(3).  It is the code which was given out at the 1985
 * UNIFORUM conference in Dallas.  I obtained it by electronic mail
 * directly from AT&T.  The people there assure me that it is indeed
 * in the public domain.
 */
 #include <stdio.h>
 #include <string.h>
 static int opterr = 1;
 static int optind = 1;
 static int optopt;
 static char *optarg;
 static int getopt(int argc, char *argv[], char *opts)
 {
    static int sp = 1;
    int c;
    char *cp;
    if (sp == 1) {
        if (optind >= argc ||
            argv[optind][0] != '-' || argv[optind][1] == '\0')
            return EOF;
        else if (!strcmp(argv[optind], "--")) {
            optind++;
            return EOF;
        }
    }
    optopt = c = argv[optind][sp];
    if (c == ':' || !(cp = strchr(opts, c))) {
        fprintf(stderr, ": illegal option -- %c\n", c);
        if (argv[optind][++sp] == '\0') {
            optind++;
            sp = 1;
        }
        return '?';
    }
    if (*++cp == ':') {
        if (argv[optind][sp+1] != '\0')
            optarg = &argv[optind++][sp+1];
        else if(++optind >= argc) {
            fprintf(stderr, ": option requires an argument -- %c\n", c);
            sp = 1;
            return '?';
        } else
            optarg = argv[optind++];
        sp = 1;
    } else {
        if (argv[optind][++sp] == '\0') {
            sp = 1;
            optind++;
        }
        optarg = NULL;
    }
    return c;
 }
--- a/compat/msvcrt/snprintf.c
+++ b/compat/msvcrt/snprintf.c
@@ -1,71 +0,0 @@
 /*
 * C99-compatible snprintf() and vsnprintf() implementations
 * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #include <stdio.h>
 #include <stdarg.h>
 #include <limits.h>
 #include <string.h>
 #include "compat/va_copy.h"
 #include "libavutil/error.h"
 #if defined(__MINGW32__)
 #define EOVERFLOW EFBIG
 #endif
 int avpriv_snprintf(char *s, size_t n, const char *fmt, ...)
 {
    va_list ap;
    int ret;
    va_start(ap, fmt);
    ret = avpriv_vsnprintf(s, n, fmt, ap);
    va_end(ap);
    return ret;
 }
 int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
                     va_list ap)
 {
    int ret;
    va_list ap_copy;
    if (n == 0)
        return _vscprintf(fmt, ap);
    else if (n > INT_MAX)
        return AVERROR(EOVERFLOW);
    /* we use n - 1 here because if the buffer is not big enough, the MS
     * runtime libraries don't add a terminating zero at the end. MSDN
     * recommends to provide _snprintf/_vsnprintf() a buffer size that
     * is one less than the actual buffer, and zero it before calling
     * _snprintf/_vsnprintf() to workaround this problem.
     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
    memset(s, 0, n);
    va_copy(ap_copy, ap);
    ret = _vsnprintf(s, n - 1, fmt, ap_copy);
    va_end(ap_copy);
    if (ret == -1)
        ret = _vscprintf(fmt, ap);
    return ret;
 }
--- a/compat/msvcrt/snprintf.h
+++ b/compat/msvcrt/snprintf.h
@@ -1,38 +0,0 @@
 /*
 * C99-compatible snprintf() and vsnprintf() implementations
 * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_SNPRINTF_H
 #define COMPAT_SNPRINTF_H
 #include <stdarg.h>
 #include <stdio.h>
 int avpriv_snprintf(char *s, size_t n, const char *fmt, ...);
 int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
 #undef snprintf
 #undef _snprintf
 #undef vsnprintf
 #define snprintf avpriv_snprintf
 #define _snprintf avpriv_snprintf
 #define vsnprintf avpriv_vsnprintf
 #endif /* COMPAT_SNPRINTF_H */
--- a/compat/os2threads.h
+++ b/compat/os2threads.h
@@ -1,166 +0,0 @@
 /*
 * Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
 *
 * 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
 * os2threads to pthreads wrapper
 */
 #ifndef AVCODEC_OS2PTHREADS_H
 #define AVCODEC_OS2PTHREADS_H
 #define INCL_DOS
 #include <os2.h>
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>
 #include "libavutil/mem.h"
 typedef TID  pthread_t;
 typedef void pthread_attr_t;
 typedef HMTX pthread_mutex_t;
 typedef void pthread_mutexattr_t;
 typedef struct {
    HEV  event_sem;
    int  wait_count;
 } pthread_cond_t;
 typedef void pthread_condattr_t;
 struct thread_arg {
    void *(*start_routine)(void *);
    void *arg;
 };
 static void thread_entry(void *arg)
 {
    struct thread_arg *thread_arg = arg;
    thread_arg->start_routine(thread_arg->arg);
    av_free(thread_arg);
 }
 static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg)
 {
    struct thread_arg *thread_arg;
    thread_arg = av_mallocz(sizeof(struct thread_arg));
    if (!thread_arg)
        return ENOMEM;
    thread_arg->start_routine = start_routine;
    thread_arg->arg = arg;
    *thread = _beginthread(thread_entry, NULL, 256 * 1024, thread_arg);
    return 0;
 }
 static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
 {
    DosWaitThread((PTID)&thread, DCWW_WAIT);
    return 0;
 }
 static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr)
 {
    DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
    return 0;
 }
 static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
 {
    DosCloseMutexSem(*(PHMTX)mutex);
    return 0;
 }
 static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
 {
    DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
    return 0;
 }
 static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
 {
    DosReleaseMutexSem(*(PHMTX)mutex);
    return 0;
 }
 static av_always_inline int pthread_cond_init(pthread_cond_t *cond, const pthread_condattr_t *attr)
 {
    DosCreateEventSem(NULL, &cond->event_sem, DCE_POSTONE, FALSE);
    cond->wait_count = 0;
    return 0;
 }
 static av_always_inline int pthread_cond_destroy(pthread_cond_t *cond)
 {
    DosCloseEventSem(cond->event_sem);
    return 0;
 }
 static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
 {
    if (cond->wait_count > 0) {
        DosPostEventSem(cond->event_sem);
        cond->wait_count--;
    }
    return 0;
 }
 static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
 {
    while (cond->wait_count > 0) {
        DosPostEventSem(cond->event_sem);
        cond->wait_count--;
    }
    return 0;
 }
 static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
    cond->wait_count++;
    pthread_mutex_unlock(mutex);
    DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT);
    pthread_mutex_lock(mutex);
    return 0;
 }
 #endif /* AVCODEC_OS2PTHREADS_H */
--- a/compat/plan9/head
+++ b/compat/plan9/head
@@ -1,10 +0,0 @@
 #!/bin/sh
 n=10
 case "$1" in
    -n)  n=$2;      shift 2 ;;
    -n*) n=${1#-n}; shift   ;;
 esac
 exec sed ${n}q "$@"
--- a/compat/plan9/main.c
+++ b/compat/plan9/main.c
@@ -1,34 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 int plan9_main(int argc, char **argv);
 #undef main
 int main(int argc, char **argv)
 {
    /* The setfcr() function in lib9 is broken, must use asm. */
 #ifdef __i386
    short fcr;
    __asm__ volatile ("fstcw        %0 \n"
                      "or      $63, %0 \n"
                      "fldcw        %0 \n"
                      : "=m"(fcr));
 #endif
    return plan9_main(argc, argv);
 }
--- a/compat/plan9/printf
+++ b/compat/plan9/printf
@@ -1,2 +0,0 @@
 #!/bin/sh
 exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"
--- a/compat/strtod.c
+++ b/compat/strtod.c
@@ -1,93 +0,0 @@
 /*
 * C99-compatible strtod() implementation
 * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #include <limits.h>
 #include <stdlib.h>
 #include "libavutil/avstring.h"
 #include "libavutil/mathematics.h"
 static char *check_nan_suffix(char *s)
 {
    char *start = s;
    if (*s++ != '(')
        return start;
    while ((*s >= 'a' && *s <= 'z') || (*s >= 'A' && *s <= 'Z') ||
           (*s >= '0' && *s <= '9') ||  *s == '_')
        s++;
    return *s == ')' ? s + 1 : start;
 }
 #undef strtod
 double strtod(const char *, char **);
 double avpriv_strtod(const char *nptr, char **endptr)
 {
    char *end;
    double res;
    /* Skip leading spaces */
    while (av_isspace(*nptr))
        nptr++;
    if (!av_strncasecmp(nptr, "infinity", 8)) {
        end = nptr + 8;
        res = INFINITY;
    } else if (!av_strncasecmp(nptr, "inf", 3)) {
        end = nptr + 3;
        res = INFINITY;
    } else if (!av_strncasecmp(nptr, "+infinity", 9)) {
        end = nptr + 9;
        res = INFINITY;
    } else if (!av_strncasecmp(nptr, "+inf", 4)) {
        end = nptr + 4;
        res = INFINITY;
    } else if (!av_strncasecmp(nptr, "-infinity", 9)) {
        end = nptr + 9;
        res = -INFINITY;
    } else if (!av_strncasecmp(nptr, "-inf", 4)) {
        end = nptr + 4;
        res = -INFINITY;
    } else if (!av_strncasecmp(nptr, "nan", 3)) {
        end = check_nan_suffix(nptr + 3);
        res = NAN;
    } else if (!av_strncasecmp(nptr, "+nan", 4) ||
               !av_strncasecmp(nptr, "-nan", 4)) {
        end = check_nan_suffix(nptr + 4);
        res = NAN;
    } else if (!av_strncasecmp(nptr, "0x", 2) ||
               !av_strncasecmp(nptr, "-0x", 3) ||
               !av_strncasecmp(nptr, "+0x", 3)) {
        /* FIXME this doesn't handle exponents, non-integers (float/double)
         * and numbers too large for long long */
        res = strtoll(nptr, &end, 16);
    } else {
        res = strtod(nptr, &end);
    }
    if (endptr)
        *endptr = end;
    return res;
 }
--- a/compat/tms470/math.h
+++ b/compat/tms470/math.h
@@ -1,30 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef FFMPEG_COMPAT_TMS470_MATH_H
 #define FFMPEG_COMPAT_TMS470_MATH_H
 #include_next <math.h>
 #undef INFINITY
 #undef NAN
 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
 #endif /* FFMPEG_COMPAT_TMS470_MATH_H */
--- a/compat/va_copy.h
+++ b/compat/va_copy.h
@@ -1,29 +0,0 @@
 /*
 * MSVC Compatible va_copy macro
 * Copyright (c) 2012 Derek Buitenhuis
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #include <stdarg.h>
 #if !defined(va_copy) && defined(_MSC_VER)
 #define va_copy(dst, src) ((dst) = (src))
 #endif
 #if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
 #define va_copy(dst, src) __va_copy(dst, src)
 #endif
--- a/compat/w32pthreads.h
+++ b/compat/w32pthreads.h
@@ -1,316 +0,0 @@
 /*
 * Copyright (C) 2010-2011 x264 project
 *
 * Authors: Steven Walters <kemuri9@gmail.com>
 *          Pegasys Inc. <http://www.pegasys-inc.com>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 /**
 * @file
 * w32threads to pthreads wrapper
 */
 #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
 * library.
 * As most functions here are used without checking return values,
 * only implement return values as necessary. */
 #define WIN32_LEAN_AND_MEAN
 #include <windows.h>
 #include <process.h>
 #include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"
 typedef struct pthread_t {
    void *handle;
    void *(*func)(void* arg);
    void *arg;
    void *ret;
 } pthread_t;
 /* the conditional variable api for windows 6.0+ uses critical sections and
 * not mutexes */
 typedef CRITICAL_SECTION pthread_mutex_t;
 /* This is the CONDITION_VARIABLE typedef for using Windows' native
 * conditional variables on kernels 6.0+. */
 #if HAVE_CONDITION_VARIABLE_PTR
 typedef CONDITION_VARIABLE pthread_cond_t;
 #else
 typedef struct pthread_cond_t {
    void *Ptr;
 } pthread_cond_t;
 #endif
 #if _WIN32_WINNT >= 0x0600
 #define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
 #define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
 #endif
 static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
    pthread_t *h = arg;
    h->ret = h->func(h->arg);
    return 0;
 }
 static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
                                    void *(*start_routine)(void*), void *arg)
 {
    thread->func   = start_routine;
    thread->arg    = arg;
    thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
                                           0, NULL);
    return !thread->handle;
 }
 static av_unused void pthread_join(pthread_t thread, void **value_ptr)
 {
    DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
    if (ret != WAIT_OBJECT_0)
        return;
    if (value_ptr)
        *value_ptr = thread.ret;
    CloseHandle(thread.handle);
 }
 static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
 {
    InitializeCriticalSection(m);
    return 0;
 }
 static inline int pthread_mutex_destroy(pthread_mutex_t *m)
 {
    DeleteCriticalSection(m);
    return 0;
 }
 static inline int pthread_mutex_lock(pthread_mutex_t *m)
 {
    EnterCriticalSection(m);
    return 0;
 }
 static inline int pthread_mutex_unlock(pthread_mutex_t *m)
 {
    LeaveCriticalSection(m);
    return 0;
 }
 #if _WIN32_WINNT >= 0x0600
 static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
    InitializeConditionVariable(cond);
    return 0;
 }
 /* native condition variables do not destroy */
 static inline void pthread_cond_destroy(pthread_cond_t *cond)
 {
    return;
 }
 static inline void pthread_cond_broadcast(pthread_cond_t *cond)
 {
    WakeAllConditionVariable(cond);
 }
 static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
    SleepConditionVariableCS(cond, mutex, INFINITE);
    return 0;
 }
 static inline void pthread_cond_signal(pthread_cond_t *cond)
 {
    WakeConditionVariable(cond);
 }
 #else // _WIN32_WINNT < 0x0600
 /* for pre-Windows 6.0 platforms we need to define and use our own condition
 * variable and api */
 typedef struct  win32_cond_t {
    pthread_mutex_t mtx_broadcast;
    pthread_mutex_t mtx_waiter_count;
    volatile int waiter_count;
    HANDLE semaphore;
    HANDLE waiters_done;
    volatile int is_broadcast;
 } win32_cond_t;
 /* function pointers to conditional variable API on windows 6.0+ kernels */
 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);
 static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
    win32_cond_t *win32_cond = NULL;
    if (cond_init) {
        cond_init(cond);
        return 0;
    }
    /* non native condition variables */
    win32_cond = av_mallocz(sizeof(win32_cond_t));
    if (!win32_cond)
        return ENOMEM;
    cond->Ptr = win32_cond;
    win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
    if (!win32_cond->semaphore)
        return ENOMEM;
    win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
    if (!win32_cond->waiters_done)
        return ENOMEM;
    pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
    pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
    return 0;
 }
 static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
 {
    win32_cond_t *win32_cond = cond->Ptr;
    /* native condition variables do not destroy */
    if (cond_init)
        return;
    /* non native condition variables */
    CloseHandle(win32_cond->semaphore);
    CloseHandle(win32_cond->waiters_done);
    pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
    pthread_mutex_destroy(&win32_cond->mtx_broadcast);
    av_freep(&win32_cond);
    cond->Ptr = NULL;
 }
 static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
 {
    win32_cond_t *win32_cond = cond->Ptr;
    int have_waiter;
    if (cond_broadcast) {
        cond_broadcast(cond);
        return;
    }
    /* non native condition variables */
    pthread_mutex_lock(&win32_cond->mtx_broadcast);
    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
    have_waiter = 0;
    if (win32_cond->waiter_count) {
        win32_cond->is_broadcast = 1;
        have_waiter = 1;
    }
    if (have_waiter) {
        ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
        ResetEvent(win32_cond->waiters_done);
        win32_cond->is_broadcast = 0;
    } else
        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
 static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
    win32_cond_t *win32_cond = cond->Ptr;
    int last_waiter;
    if (cond_wait) {
        cond_wait(cond, mutex, INFINITE);
        return 0;
    }
    /* non native condition variables */
    pthread_mutex_lock(&win32_cond->mtx_broadcast);
    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
    win32_cond->waiter_count++;
    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
    // unlock the external mutex
    pthread_mutex_unlock(mutex);
    WaitForSingleObject(win32_cond->semaphore, INFINITE);
    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
    win32_cond->waiter_count--;
    last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
    if (last_waiter)
        SetEvent(win32_cond->waiters_done);
    // lock the external mutex
    return pthread_mutex_lock(mutex);
 }
 static av_unused void pthread_cond_signal(pthread_cond_t *cond)
 {
    win32_cond_t *win32_cond = cond->Ptr;
    int have_waiter;
    if (cond_signal) {
        cond_signal(cond);
        return;
    }
    pthread_mutex_lock(&win32_cond->mtx_broadcast);
    /* non-native condition variables */
    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
    have_waiter = win32_cond->waiter_count;
    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
    if (have_waiter) {
        ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
        ResetEvent(win32_cond->waiters_done);
    }
    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
 #endif
 static av_unused 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      =
        (void*)GetProcAddress(kernel_dll, "InitializeConditionVariable");
    cond_broadcast =
        (void*)GetProcAddress(kernel_dll, "WakeAllConditionVariable");
    cond_signal    =
        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
    cond_wait      =
        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
 #endif
 }
 #endif /* FFMPEG_COMPAT_W32PTHREADS_H */
--- a/compat/windows/makedef
+++ b/compat/windows/makedef
@@ -1,132 +0,0 @@
 #!/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 "${list}" | sort | uniq | tail -n +2
--- a/5692
+++ b/5692
--- a/doc/APIchanges
+++ b/doc/APIchanges
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -1,184 +0,0 @@
 LIBRARIES-$(CONFIG_AVUTIL)     += libavutil
 LIBRARIES-$(CONFIG_SWSCALE)    += libswscale
 LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
 LIBRARIES-$(CONFIG_AVCODEC)    += libavcodec
 LIBRARIES-$(CONFIG_AVFORMAT)   += libavformat
 LIBRARIES-$(CONFIG_AVDEVICE)   += libavdevice
 LIBRARIES-$(CONFIG_AVFILTER)   += libavfilter
 COMPONENTS-$(CONFIG_AVUTIL)     += ffmpeg-utils
 COMPONENTS-$(CONFIG_SWSCALE)    += ffmpeg-scaler
 COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
 COMPONENTS-$(CONFIG_AVCODEC)    += ffmpeg-codecs ffmpeg-bitstream-filters
 COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters
 MANPAGES1   = $(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) $(LIBRARIES-yes:%=doc/%.html) \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
              doc/general.html                                          \
              doc/git-howto.html                                        \
              doc/nut.html                                              \
              doc/platform.html                                         \
 TXTPAGES    = doc/fate.txt                                              \
 DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
 DOCS-$(CONFIG_PODPAGES)  += $(PODPAGES)
 DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 DOC_EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
 DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
 DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
 DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
 DOC_EXAMPLES-$(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-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
 DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
 DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
 DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
 DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
 DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
 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
 apidoc: doc/doxy/html
 documentation: $(DOCS)
 examples: $(DOC_EXAMPLES)
 TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
 GENTEXI  = format codec
 GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
 $(GENTEXI): TAG = GENTEXI
 $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 	$(M)doc/print_options $* > $@
 doc/%.html: TAG = HTML
 doc/%-all.html: TAG = HTML
 ifdef HAVE_MAKEINFO_HTML
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
 doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
 else
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 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 $@ $<
 endif
 doc/%.pod: TAG = POD
 doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(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)
 	$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
 doc/%.3: doc/%.pod $(GENTEXI)
 	$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
 $(DOCS) doc/doxy/html: | doc/
 $(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
 OBJDIRS += doc/examples
 DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
 doc/doxy/html: TAG = DOXY
 doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
 	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
 install-doc: install-html install-man
 install-html:
 install-man:
 ifdef CONFIG_HTMLPAGES
 install-progs-$(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 $(MANPAGES1) "$(MANDIR)/man1"
 	$(Q)mkdir -p "$(MANDIR)/man3"
 	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
 endif
 uninstall: uninstall-doc
 uninstall-doc: uninstall-html uninstall-man
 uninstall-html:
 	$(RM) -r "$(DOCDIR)"
 uninstall-man:
 	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
 	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
 clean:: docclean
 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))
 .PHONY: apidoc doc documentation
--- a/doc/TODO
+++ b/doc/TODO
@@ -0,0 +1,90 @@
 ffmpeg TODO list:
 ----------------
 Fabrice's TODO list: (unordered)
 -------------------
 Short term:
 - use AVFMTCTX_DISCARD_PKT in ffplay so that DV has a chance to work
 - add RTSP regression test (both client and server)
 - make ffserver allocate AVFormatContext
 - clean up (incompatible change, for 0.5.0):
    * AVStream -> AVComponent
    * AVFormatContext -> AVInputStream/AVOutputStream
    * suppress rate_emu from AVCodecContext
 - add new float/integer audio filterting and conversion : suppress
  CODEC_ID_PCM_xxc and use CODEC_ID_RAWAUDIO.
 - fix telecine and frame rate conversion
 Long term (ask me if you want to help):
 - commit new imgconvert API and new PIX_FMT_xxx alpha formats
 - commit new LGPL'ed float and integer-only AC3 decoder
 - add WMA integer-only decoder
 - add new MPEG4-AAC audio decoder (both integer-only and float version)
 Michael's TODO list: (unordered) (if anyone wanna help with sth, just ask)
 -------------------
 - optimize H264 CABAC
 - more optimizations
 - simper rate control
 Francois' TODO list: (unordered, without any timeframe)
 -------------------
 - test MACE decoder against the openquicktime one as suggested by A'rpi
 - BeOS audio input grabbing backend
 - BeOS video input grabbing backend
 - publish my BeOS libposix on BeBits so I can officially support ffserver :)
 - check the whole code for thread-safety (global and init stuff)
 Philip'a TODO list: (alphabetically ordered) (please help)
 ------------------
 - Add a multi-ffm filetype so that feeds can be recorded into multiple files rather
  than one big file.
 - Authenticated users support -- where the authentication is in the URL
 - Change ASF files so that the embedded timestamp in the frames is right rather
  than being an offset from the start of the stream
 - Make ffm files more resilient to changes in the codec structures so that you
  can play old ffm files.
 Baptiste's TODO list:
 -----------------
 - mov edit list support (AVEditList)
 - YUV 10 bit per component support "2vuy"
 - mxf muxer
 - mpeg2 non linear quantizer
 unassigned TODO: (unordered)
 ---------------
 - use AVFrame for audio codecs too
 - rework aviobuf.c buffering strategy and fix url_fskip
 - generate optimal huffman tables for mjpeg encoding
 - fix ffserver regression tests
 - support xvids motion estimation
 - support x264s motion estimation
 - support x264s rate control
 - SNOW: non translational motion compensation
 - SNOW: more optimal quantization
 - SNOW: 4x4 block support
 - SNOW: 1/8 pel motion compensation support
 - SNOW: iterative motion estimation based on subsampled images
 - SNOW: try B frames and MCTF and see how their PSNR/bitrate/complexity behaves
 - SNOW: try to use the wavelet transformed MC-ed reference frame as context for the entropy coder
 - SNOW: think about/analyize how to make snow use multiple cpus/threads
 - SNOW: finish spec
 - FLAC: lossy encoding (viterbi and naive scalar quantization)
 - libavfilter
 - JPEG2000 decoder & encoder
 - MPEG4 GMC encoding support
 - macroblock based pixel format (better cache locality, somewhat complex, one paper claimed it faster for high res)
 - regression tests for codecs which do not have an encoder (I+P-frame bitstream in svn)
 - add support for using mplayers video filters to ffmpeg
 - H264 encoder
 - per MB ratecontrol (so VCD and such do work better)
 - write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions
 - convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc
 - generic audio mixing API
 - extract PES packetizer from PS muxer and use it for new TS muxer
 - implement automatic AVBistreamFilter activation
 - make cabac encoder use bytestream (see http://trac.videolan.org/x264/changeset/?format=diff&new=651)
 - merge imdct and windowing, the current code does considerable amounts of redundant work
--- a/doc/authors.texi
+++ b/doc/authors.texi
@@ -1,11 +0,0 @@
@chapter Authors
 The FFmpeg developers.
 For details about the authorship, see the Git history of the project
 (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
@command{git log} in the FFmpeg source directory, or browsing the
 online repository at @url{http://source.ffmpeg.org}.
 Maintainers for the specific components are listed in the file
@file{MAINTAINERS} in the source code tree.
--- a/doc/avutil.txt
+++ b/doc/avutil.txt
@@ -0,0 +1,37 @@
 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
 softfloat.c
 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)
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -1,180 +0,0 @@
@chapter Bitstream Filters
@c man begin BITSTREAM FILTERS
 When you configure your FFmpeg build, all the supported bitstream
 filters are enabled by default. You can list all available ones using
 the configure option @code{--list-bsfs}.
 You can disable all the bitstream filters using the configure option
@code{--disable-bsfs}, and selectively enable any bitstream filter using
 the option @code{--enable-bsf=BSF}, or you can disable a particular
 bitstream filter using the option @code{--disable-bsf=BSF}.
 The option @code{-bsfs} of the ff* tools will display the list of
 all the supported bitstream filters included in your build.
 The ff* tools have a -bsf option applied per stream, taking a
 comma-separated list of filters, whose parameters follow the filter
 name after a '='.
@example
 ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
@end example
 Below is a description of the currently available bitstream filters,
 with their parameters, if any.
@section aac_adtstoasc
 Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
 bitstream filter.
 This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
 ADTS header and removes the ADTS header.
 This is required for example when copying an AAC stream from a raw
 ADTS AAC container to a FLV or a MOV/MP4 file.
@section chomp
 Remove zero padding at the end of a packet.
@section dump_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
 Convert an H.264 bitstream from length prefixed mode to start code
 prefixed mode (as defined in the Annex B of the ITU-T H.264
 specification).
 This is required by some streaming formats, typically the MPEG-2
 transport stream format ("mpegts").
 For example to remux an MP4 file containing an H.264 stream to mpegts
 format with @command{ffmpeg}, you can use the command:
@example
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
@section imxdump
 Modifies the bitstream to fit in MOV and to be usable by the Final Cut
 Pro decoder. This filter only applies to the mpeg2video codec, and is
 likely not needed for Final Cut Pro 7 and newer with the appropriate
@option{-tag:v}.
 For example, to remux 30 MB/sec NTSC IMX to MOV:
@example
 ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
@end example
@section mjpeg2jpeg
 Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
 MJPEG is a video codec wherein each video frame is essentially a
 JPEG image. The individual frames can be extracted without loss,
 e.g. by
@example
 ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
@end example
 Unfortunately, these chunks are incomplete JPEG images, because
 they lack the DHT segment required for decoding. Quoting from
@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
 Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
 commented that "MJPEG, or at least the MJPEG in AVIs having the
 MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
 Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
 and it must use basic Huffman encoding, not arithmetic or
 progressive. . . . You can indeed extract the MJPEG frames and
 decode them with a regular JPEG decoder, but you have to prepend
 the DHT segment to them, or else the decoder won't have any idea
 how to decompress the data. The exact table necessary is given in
 the OpenDML spec."
 This bitstream filter patches the header of frames extracted from an MJPEG
 stream (carrying the AVI1 header ID and lacking a DHT segment) to
 produce fully qualified JPEG images.
@example
 ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
 exiftran -i -9 frame*.jpg
 ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpega_dump_header
@section movsub
@section mp3_header_decompress
@section mpeg4_unpack_bframes
 Unpack DivX-style packed B-frames.
 DivX-style packed B-frames are not valid MPEG-4 and were only a
 workaround for the broken Video for Windows subsystem.
 They use more space, can cause minor AV sync issues, require more
 CPU power to decode (unless the player has some decoded picture queue
 to compensate the 2,0,2,0 frame per packet style) and cause
 trouble if copied into a standard container like mp4 or mpeg-ps/ts,
 because MPEG-4 decoders may not be able to decode them, since they are
 not valid MPEG-4.
 For example to fix an AVI file containing an MPEG-4 stream with
 DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
@example
 ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@end example
@section noise
 Damages the contents of packets without damaging the container. Can be
 used for fuzzing or testing error resilience/concealment.
 Parameters:
 A numeral string, whose value is related to how often output bytes will
 be modified. Therefore, values below or equal to 0 are forbidden, and
 the lower the more frequent bytes will be modified, with 1 meaning
 every byte is modified.
@example
 ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@end example
 applies the modification to every byte.
@section remove_extra
@c man end BITSTREAM FILTERS
--- a/doc/bootstrap.min.css
+++ b/doc/bootstrap.min.css
--- a/doc/build_system.txt
+++ b/doc/build_system.txt
@@ -1,64 +0,0 @@
 FFmpeg currently uses a custom build system, this text attempts to document
 some of its obscure features and options.
 Makefile variables:
 V
    Disable the default terse mode, the full command issued by make and its
    output will be shown on the screen.
 DBG
    Preprocess x86 external assembler files to a .dbg.asm file in the object
    directory, which then gets compiled. Helps developping those assembler
    files.
 DESTDIR
    Destination directory for the install targets, useful to prepare packages
    or install FFmpeg in cross-environments.
 GEN
    Set to ‘1’ to generate the missing or mismatched references.
 Makefile targets:
 all
    Default target, builds all the libraries and the executables.
 fate
    Run the fate test suite, note you must have installed it
 fate-list
    Will list all fate/regression test targets
 install
    Install headers, libraries and programs.
 examples
    Build all examples located in doc/examples.
 libavformat/output-example
    Build the libavformat basic example.
 libavcodec/api-example
    Build the libavcodec basic example.
 libswscale/swscale-test
    Build the swscale self-test (useful also as example).
 config
    Reconfigure the project with current configuration.
 Useful standard make commands:
 make -t <target>
    Touch all files that otherwise would be build, this is useful to reduce
    unneeded rebuilding when changing headers, but note you must force rebuilds
    of files that actually need it by hand then.
 make -j<num>
    rebuild with multiple jobs at the same time. Faster on multi processor systems
 make -k
    continue build in case of errors, this is useful for the regression tests
    sometimes but note it will still not run all reg tests.
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -1,262 +0,0 @@
@chapter Decoders
@c man begin DECODERS
 Decoders are configured elements in FFmpeg which allow the decoding of
 multimedia streams.
 When you configure your FFmpeg build, all the supported native decoders
 are enabled by default. Decoders requiring an external library must be enabled
 manually via the corresponding @code{--enable-lib} option. You can list all
 available decoders using the configure option @code{--list-decoders}.
 You can disable all the decoders with the configure option
@code{--disable-decoders} and selectively enable / disable single decoders
 with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
 The option @code{-decoders} of the ff* tools will display the list of
 enabled decoders.
@c man end DECODERS
@chapter Video Decoders
@c man begin VIDEO DECODERS
 A description of some of the currently available video decoders
 follows.
@section rawvideo
 Raw video decoder.
 This decoder decodes rawvideo streams.
@subsection Options
@table @option
@item top @var{top_field_first}
 Specify the assumed field type of the input video.
@table @option
@item -1
 the video is assumed to be progressive (default)
@item 0
 bottom-field-first is assumed
@item 1
 top-field-first is assumed
@end table
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@c man begin AUDIO DECODERS
 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 flac
 FLAC audio decoder.
 This decoder aims to implement the complete FLAC specification from Xiph.
@subsection FLAC Decoder options
@table @option
@item -use_buggy_lpc
 The lavc FLAC encoder used to produce buggy streams with high lpc values
 (like the default value). This option makes it possible to decode such streams
 correctly by using lavc's old buggy lpc logic for decoding.
@end table
@section ffwavesynth
 Internal wave synthetizer.
 This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
 documented.
@section libcelt
 libcelt decoder wrapper.
 libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
 Requires the presence of the libcelt headers and library during configuration.
 You need to explicitly configure the build with @code{--enable-libcelt}.
@section libgsm
 libgsm decoder wrapper.
 libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
 the presence of the libgsm headers and library during configuration. You need
 to explicitly configure the build with @code{--enable-libgsm}.
 This decoder supports both the ordinary GSM and the Microsoft variant.
@section libilbc
 libilbc decoder wrapper.
 libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
 audio codec. Requires the presence of the libilbc headers and library during
 configuration. You need to explicitly configure the build with
@code{--enable-libilbc}.
@subsection Options
 The following option is supported by the libilbc wrapper.
@table @option
@item enhance
 Enable the enhancement of the decoded audio when set to 1. The default
 value is 0 (disabled).
@end table
@section libopencore-amrnb
 libopencore-amrnb decoder wrapper.
 libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
 Narrowband audio codec. Using it requires the presence of the
 libopencore-amrnb headers and library during configuration. You need to
 explicitly configure the build with @code{--enable-libopencore-amrnb}.
 An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
 without this library.
@section libopencore-amrwb
 libopencore-amrwb decoder wrapper.
 libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
 Wideband audio codec. Using it requires the presence of the
 libopencore-amrwb headers and library during configuration. You need to
 explicitly configure the build with @code{--enable-libopencore-amrwb}.
 An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
 without this library.
@section libopus
 libopus decoder wrapper.
 libopus allows libavcodec to decode the Opus Interactive Audio Codec.
 Requires the presence of the libopus headers and library during
 configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
 An FFmpeg native decoder for Opus exists, so users can decode Opus
 without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section dvdsub
 This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
 also be found in VobSub file pairs and in some Matroska files.
@subsection Options
@table @option
@item palette
 Specify the global palette used by the bitmaps. When stored in VobSub, the
 palette is normally specified in the index file; in Matroska, the palette is
 stored in the codec extra-data in the same format as in VobSub. In DVDs, the
 palette is stored in the IFO file, and therefore not available when reading
 from dumped VOB files.
 The format for this option is a string containing 16 24-bits hexadecimal
 numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
 ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@item ifo_palette
 Specify the IFO file from which the global palette is obtained.
 (experimental)
@item forced_subs_only
 Only decode subtitle entries marked as forced. Some titles have forced
 and non-forced subtitles in the same track. Setting this flag to @code{1}
 will only keep the forced subtitles. Default value is @code{0}.
@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
--- a/doc/default.css
+++ b/doc/default.css
@@ -1,165 +0,0 @@
 a.summary-letter {
    text-decoration: none;
 }
 a {
    color: #2D6198;
 }
 a:visited {
    color: #884488;
 }
 #banner {
    background-color: white;
    position: relative;
    text-align: center;
 }
 #banner img {
    margin-bottom: 1px;
    margin-top: 5px;
 }
 #body {
    margin-left: 1em;
    margin-right: 1em;
 }
 body {
    background-color: #313131;
    margin: 0;
    text-align: justify;
 }
 .center {
    margin-left: auto;
    margin-right: auto;
    text-align: center;
 }
 #container {
    background-color: white;
    color: #202020;
    margin-left: 1em;
    margin-right: 1em;
 }
 #footer {
    text-align: center;
 }
 h1 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: 0.25em;
    padding-top: 0.25em;
    border: 1px solid #6A996A;
 }
 h1 {
    background-color: #7BB37B;
    color: #151515;
    font-size: 1.2em;
    padding-bottom: 0.3em;
    padding-top: 0.3em;
 }
 h2 {
    color: #313131;
    font-size: 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: #D1FDD1;
 }
 img {
    border: 0;
 }
 #navbar {
    background-color: #738073;
    border-bottom: 1px solid #5C665C;
    border-top: 1px solid #5C665C;
    margin-top: 12px;
    padding: 0.3em;
    position: relative;
    text-align: center;
 }
 #navbar a, #navbar_secondary a {
    color: white;
    padding: 0.3em;
    text-decoration: none;
 }
 #navbar a:hover, #navbar_secondary a:hover {
    background-color: #313131;
    color: white;
    text-decoration: none;
 }
 #navbar_secondary {
    background-color: #738073;
    border-bottom: 1px solid #5C665C;
    border-left: 1px solid #5C665C;
    border-right: 1px solid #5C665C;
    padding: 0.3em;
    position: relative;
    text-align: center;
 }
 p {
    margin-left: 1em;
    margin-right: 1em;
 }
 pre {
    margin-left: 3em;
    margin-right: 3em;
    padding: 0.3em;
    border: 1px solid #bbb;
    background-color: #f7f7f7;
 }
 dl dt {
    font-weight: bold;
 }
 #proj_desc {
    font-size: 1.2em;
 }
 #repos {
    margin-left: 1em;
    margin-right: 1em;
    border-collapse: collapse;
    border: solid 1px #6A996A;
 }
 #repos th {
    background-color: #7BB37B;
    border: solid 1px #6A996A;
 }
 #repos td {
    padding: 0.2em;
    border: solid 1px #6A996A;
 }
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -1,465 +0,0 @@
@chapter Demuxers
@c man begin DEMUXERS
 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
 are enabled by default. You can list all available ones using the
 configure option @code{--list-demuxers}.
 You can disable all the demuxers using the configure option
@code{--disable-demuxers}, and selectively enable a single demuxer with
 the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
 with the option @code{--disable-demuxer=@var{DEMUXER}}.
 The option @code{-formats} of the ff* tools will display the list of
 enabled demuxers.
 The description of some of the currently available demuxers follows.
@section applehttp
 Apple HTTP Live Streaming demuxer.
 This demuxer presents all AVStreams from all variant streams.
 The id field is set to the bitrate variant index number. By setting
 the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
 the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".
@section apng
 Animated Portable Network Graphics demuxer.
 This demuxer is used to demux APNG files.
 All headers, but the PNG signature, up to (but not including) the first
 fcTL chunk are transmitted as extradata.
 Frames are then split as being all the chunks between two fcTL ones, or
 between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
 Ignore the loop variable in the file if set.
@item -max_fps @var{int}
 Maximum framerate in frames per second (0 for no limit).
@item -default_fps @var{int}
 Default framerate in frames per second when none is specified in the file
 (0 meaning as fast as possible).
@end table
@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
 Virtual concatenation script demuxer.
 This demuxer reads a list of files and other directives from a text file and
 demuxes them one after the other, as if all their packet had been muxed
 together.
 The timestamps in the files are adjusted so that the first file starts at 0
 and each next file starts where the previous one finishes. Note that it is
 done globally and may cause gaps if all streams do not have exactly the same
 length.
 All files must have the same streams (same codecs, same time base, etc.).
 The duration of each file is used to adjust the timestamps of the next file:
 if the duration is incorrect (because it was computed using the bit-rate or
 because the file is truncated, for example), it can cause artifacts. The
@code{duration} directive can be used to override the duration stored in
 each file.
@subsection Syntax
 The script is a text file in extended-ASCII, with one directive per line.
 Empty lines, leading spaces and lines starting with '#' are ignored. The
 following directive is recognized:
@table @option
@item @code{file @var{path}}
 Path to a file to read; special characters and spaces must be escaped with
 backslash or single quotes.
 All subsequent file-related directives apply to that file.
@item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
 to 1 if it was to its default -1.
 To make FFmpeg recognize the format automatically, this directive must
 appears exactly as is (no extra space or byte-order-mark) on the very first
 line of the script.
@item @code{duration @var{dur}}
 Duration of the file. This information can be specified from the file;
 specifying it here may be more efficient or help if the information from the
 file is not available or accurate.
 If the duration is set for all files, then it is possible to seek in the
 whole concatenated video.
@item @code{stream}
 Introduce a stream in the virtual file.
 All subsequent stream-related directives apply to the last introduced
 stream.
 Some streams properties must be set in order to allow identifying the
 matching streams in the subfiles.
 If no streams are defined in the script, the streams from the first file are
 copied.
@item @code{exact_stream_id @var{id}}
 Set the id of the stream.
 If this directive is given, the string with the corresponding id in the
 subfiles will be used.
 This is especially useful for MPEG-PS (VOB) files, where the order of the
 streams is not reliable.
@end table
@subsection Options
 This demuxer accepts the following option:
@table @option
@item safe
 If set to 1, reject unsafe file paths. A file path is considered safe if it
 does not contain a protocol specification and is relative and all components
 only contain characters from the portable character set (letters, digits,
 period, underscore and hyphen) and have no period at the beginning of a
 component.
 If set to 0, any file name is accepted.
 The default is -1, it is equivalent to 1 if the format was automatically
 probed and 0 otherwise.
@item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
 The default is 1.
 Currently, the only conversion is adding the h264_mp4toannexb bitstream
 filter to H.264 streams in MP4 format. This is necessary in particular if
 there are resolution changes.
@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 gif
 Animated GIF demuxer.
 It accepts the following options:
@table @option
@item min_delay
 Set the minimum valid delay between frames in hundredths of seconds.
 Range is 0 to 6000. Default value is 2.
@item max_gif_delay
 Set the maximum valid delay between frames in hundredth of seconds.
 Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
 the maximum value allowed by the specification.
@item default_delay
 Set the default delay between frames in hundredths of seconds.
 Range is 0 to 6000. Default value is 10.
@item ignore_loop
 GIF files can contain information to loop a certain number of times (or
 infinitely). If @option{ignore_loop} is set to 1, then the loop setting
 from the input will be ignored and looping will not occur. If set to 0,
 then looping will occur and will cycle the number of times according to
 the GIF. Default value is 1.
@end table
 For example, with the overlay filter, place an infinitely looping GIF
 over another video:
@example
 ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
 Note that in the above example the shortest option for overlay filter is
 used to end the output video at the length of the shortest input file,
 which in this case is @file{input.mp4} as the GIF in this example loops
 infinitely.
@section image2
 Image file demuxer.
 This demuxer reads from a list of image files specified by a pattern.
 The syntax and meaning of the pattern is specified by the
 option @var{pattern_type}.
 The pattern may contain a suffix which is used to automatically
 determine the format of the images contained in the files.
 The size, the pixel format, and the format of each image must be the
 same for all the files in the sequence.
 This demuxer accepts the following options:
@table @option
@item framerate
 Set the 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
 Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item none
 Disable pattern matching, therefore the video will only contain the specified
 image. You should use this option if you do not want to create sequences from
 multiple images and your filenames may contain special pattern characters.
@item sequence
 Select a sequence pattern type, used to specify a sequence of files
 indexed by sequential numbers.
 A sequence pattern may contain the string "%d" or "%0@var{N}d", which
 specifies the position of the characters representing a sequential
 number in each filename matched by the pattern. If the form
 "%d0@var{N}d" is used, the string representing the number in each
 filename is 0-padded and @var{N} is the total number of 0-padded
 digits representing the number. The literal character '%' can be
 specified in the pattern with the string "%%".
 If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
 the file list specified by the pattern must contain a number
 inclusively contained between @var{start_number} and
@var{start_number}+@var{start_number_range}-1, and all the following
 numbers must be sequential.
 For example the pattern "img-%03d.bmp" will match a sequence of
 filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
 sequence of filenames of the form @file{i%m%g-1.jpg},
@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
 Note that the pattern must not necessarily contain "%d" or
 "%0@var{N}d", for example to convert a single image file
@file{img.jpeg} you can employ the command:
@example
 ffmpeg -i img.jpeg img.png
@end example
@item glob
 Select a glob wildcard pattern type.
 The pattern is interpreted like a @code{glob()} pattern. This is only
 selectable if libavformat was compiled with globbing support.
@item glob_sequence @emph{(deprecated, will be removed)}
 Select a mixed glob wildcard/sequence pattern.
 If your version of libavformat was compiled with globbing support, and
 the provided pattern contains at least one glob meta character among
@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
 interpreted like a @code{glob()} pattern, otherwise it is interpreted
 like a sequence pattern.
 All glob special characters @code{%*?[]@{@}} must be prefixed
 with "%". To escape a literal "%" you shall use "%%".
 For example the pattern @code{foo-%*.jpeg} will match all the
 filenames prefixed by "foo-" and terminating with ".jpeg", and
@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
 "foo-", followed by a sequence of three characters, and terminating
 with ".jpeg".
 This pattern type is deprecated in favor of @var{glob} and
@var{sequence}.
@end table
 Default value is @var{glob_sequence}.
@item pixel_format
 Set the pixel format of the images to read. If not specified the pixel
 format is guessed from the first image file in the sequence.
@item start_number
 Set the index of the file matched by the image file pattern to start
 to read from. Default value is 0.
@item start_number_range
 Set the index interval range to check when looking for the first image
 file in the sequence, starting from @var{start_number}. Default value
 is 5.
@item 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.
 If set to 2, will set frame timestamp to the modification time of the image file in
 nanosecond precision.
@item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
@end table
@subsection Examples
@itemize
@item
 Use @command{ffmpeg} for creating a video from the images in the file
 sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
 input frame rate of 10 frames per second:
@example
 ffmpeg -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 -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 -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 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.
 This demuxer accepts the following options:
@table @option
@item framerate
 Set input video frame rate. Default value is 25.
@item pixel_format
 Set the input video pixel format. Default value is @code{yuv420p}.
@item video_size
 Set the input video size. This value must be specified explicitly.
@end table
 For example to read a rawvideo file @file{input.raw} with
@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
 size of @code{320x240}, and a frame rate of 10 images per second, use
 the command:
@example
 ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
@end example
@section sbg
 SBaGen script demuxer.
 This demuxer reads the script language used by SBaGen
@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
 script looks like that:
@example
 -SE
 a: 300-2.5/3 440+4.5/0
 b: 300-2.5/0 440+4.5/3
 off: -
 NOW      == a
 +0:07:00 == b
 +0:14:00 == a
 +0:21:00 == b
 +0:30:00    off
@end example
 A SBG script can mix absolute and relative timestamps. If the script uses
 either only absolute timestamps (including the script start time) or only
 relative ones, then its layout is fixed, and the conversion is
 straightforward. On the other hand, if the script mixes both kind of
 timestamps, then the @var{NOW} reference for relative timestamps will be
 taken from the current time of day at the time the script is read, and the
 script layout will be frozen according to that reference. That means that if
 the script is directly played, the actual times will match the absolute
 timestamps up to the sound controller's clock accuracy, but if the user
 somehow pauses the playback or seeks, all times will be shifted accordingly.
@section tedcaptions
 JSON captions used for @url{http://www.ted.com/, TED Talks}.
 TED does not provide links to the captions, but they can be guessed from the
 page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
 contains a bookmarklet to expose them.
 This demuxer accepts the following option:
@table @option
@item start_time
 Set the start time of the TED talk, in milliseconds. The default is 15000
 (15s). It is used to sync the captions with the downloadable videos, because
 they include a 15s intro.
@end table
 Example: convert the captions to a format most players understand:
@example
 ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@c man end DEMUXERS
--- a/doc/developer.texi
+++ b/doc/developer.texi
--- a/doc/devices.texi
+++ b/doc/devices.texi
@@ -1,25 +0,0 @@
@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
--- a/doc/doxy-wrapper.sh
+++ b/doc/doxy-wrapper.sh
@@ -1,21 +0,0 @@
 #!/bin/sh
 SRC_PATH="${1}"
 DOXYFILE="${2}"
 DOXYGEN="${3}"
 shift 3
 if [ -e "$SRC_PATH/VERSION" ]; then
    VERSION=`cat "$SRC_PATH/VERSION"`
 else
    VERSION=`cd "$SRC_PATH"; git describe`
 fi
 $DOXYGEN - <<EOF
@INCLUDE        = ${DOXYFILE}
 INPUT           = $@
 EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
 HTML_TIMESTAMP  = NO
 PROJECT_NUMBER  = $VERSION
 EOF
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
--- a/doc/errno.txt
+++ b/doc/errno.txt
@@ -1,174 +0,0 @@
 The following table lists most error codes found in various operating
 systems supported by FFmpeg.
                             OS
 Code             Std    F  LBMWwb Text (YMMV)
 E2BIG            POSIX     ++++++  Argument list too long
 EACCES           POSIX     ++++++  Permission denied
 EADDRINUSE       POSIX     +++..+  Address in use
 EADDRNOTAVAIL    POSIX     +++..+  Cannot assign requested address
 EADV                       +.....  Advertise error
 EAFNOSUPPORT     POSIX     +++..+  Address family not supported
 EAGAIN           POSIX  +  ++++++  Resource temporarily unavailable
 EALREADY         POSIX     +++..+  Operation already in progress
 EAUTH                      .++...  Authentication error
 EBADARCH                   ..+...  Bad CPU type in executable
 EBADE                      +.....  Invalid exchange
 EBADEXEC                   ..+...  Bad executable
 EBADF            POSIX     ++++++  Bad file descriptor
 EBADFD                     +.....  File descriptor in bad state
 EBADMACHO                  ..+...  Malformed Macho file
 EBADMSG          POSIX     ++4...  Bad message
 EBADR                      +.....  Invalid request descriptor
 EBADRPC                    .++...  RPC struct is bad
 EBADRQC                    +.....  Invalid request code
 EBADSLT                    +.....  Invalid slot
 EBFONT                     +.....  Bad font file format
 EBUSY            POSIX  -  ++++++  Device or resource busy
 ECANCELED        POSIX     +++...  Operation canceled
 ECHILD           POSIX     ++++++  No child processes
 ECHRNG                     +.....  Channel number out of range
 ECOMM                      +.....  Communication error on send
 ECONNABORTED     POSIX     +++..+  Software caused connection abort
 ECONNREFUSED     POSIX  -  +++ss+  Connection refused
 ECONNRESET       POSIX     +++..+  Connection reset
 EDEADLK          POSIX     ++++++  Resource deadlock avoided
 EDEADLOCK                  +..++.  File locking deadlock error
 EDESTADDRREQ     POSIX     +++...  Destination address required
 EDEVERR                    ..+...  Device error
 EDOM             C89    -  ++++++  Numerical argument out of domain
 EDOOFUS                    .F....  Programming error
 EDOTDOT                    +.....  RFS specific error
 EDQUOT           POSIX     +++...  Disc quota exceeded
 EEXIST           POSIX     ++++++  File exists
 EFAULT           POSIX  -  ++++++  Bad address
 EFBIG            POSIX  -  ++++++  File too large
 EFTYPE                     .++...  Inappropriate file type or format
 EHOSTDOWN                  +++...  Host is down
 EHOSTUNREACH     POSIX     +++..+  No route to host
 EHWPOISON                  +.....  Memory page has hardware error
 EIDRM            POSIX     +++...  Identifier removed
 EILSEQ           C99       ++++++  Illegal byte sequence
 EINPROGRESS      POSIX  -  +++ss+  Operation in progress
 EINTR            POSIX  -  ++++++  Interrupted system call
 EINVAL           POSIX  +  ++++++  Invalid argument
 EIO              POSIX  +  ++++++  I/O error
 EISCONN          POSIX     +++..+  Socket is already connected
 EISDIR           POSIX     ++++++  Is a directory
 EISNAM                     +.....  Is a named type file
 EKEYEXPIRED                +.....  Key has expired
 EKEYREJECTED               +.....  Key was rejected by service
 EKEYREVOKED                +.....  Key has been revoked
 EL2HLT                     +.....  Level 2 halted
 EL2NSYNC                   +.....  Level 2 not synchronized
 EL3HLT                     +.....  Level 3 halted
 EL3RST                     +.....  Level 3 reset
 ELIBACC                    +.....  Can not access a needed shared library
 ELIBBAD                    +.....  Accessing a corrupted shared library
 ELIBEXEC                   +.....  Cannot exec a shared library directly
 ELIBMAX                    +.....  Too many shared libraries
 ELIBSCN                    +.....  .lib section in a.out corrupted
 ELNRNG                     +.....  Link number out of range
 ELOOP            POSIX     +++..+  Too many levels of symbolic links
 EMEDIUMTYPE                +.....  Wrong medium type
 EMFILE           POSIX     ++++++  Too many open files
 EMLINK           POSIX     ++++++  Too many links
 EMSGSIZE         POSIX     +++..+  Message too long
 EMULTIHOP        POSIX     ++4...  Multihop attempted
 ENAMETOOLONG     POSIX  -  ++++++  Filen ame too long
 ENAVAIL                    +.....  No XENIX semaphores available
 ENEEDAUTH                  .++...  Need authenticator
 ENETDOWN         POSIX     +++..+  Network is down
 ENETRESET        SUSv3     +++..+  Network dropped connection on reset
 ENETUNREACH      POSIX     +++..+  Network unreachable
 ENFILE           POSIX     ++++++  Too many open files in system
 ENOANO                     +.....  No anode
 ENOATTR                    .++...  Attribute not found
 ENOBUFS          POSIX  -  +++..+  No buffer space available
 ENOCSI                     +.....  No CSI structure available
 ENODATA          XSR       +N4...  No message available
 ENODEV           POSIX  -  ++++++  No such device
 ENOENT           POSIX  -  ++++++  No such file or directory
 ENOEXEC          POSIX     ++++++  Exec format error
 ENOFILE                    ...++.  No such file or directory
 ENOKEY                     +.....  Required key not available
 ENOLCK           POSIX     ++++++  No locks available
 ENOLINK          POSIX     ++4...  Link has been severed
 ENOMEDIUM                  +.....  No medium found
 ENOMEM           POSIX     ++++++  Not enough space
 ENOMSG           POSIX     +++..+  No message of desired type
 ENONET                     +.....  Machine is not on the network
 ENOPKG                     +.....  Package not installed
 ENOPROTOOPT      POSIX     +++..+  Protocol not available
 ENOSPC           POSIX     ++++++  No space left on device
 ENOSR            XSR       +N4...  No STREAM resources
 ENOSTR           XSR       +N4...  Not a STREAM
 ENOSYS           POSIX  +  ++++++  Function not implemented
 ENOTBLK                    +++...  Block device required
 ENOTCONN         POSIX     +++..+  Socket is not connected
 ENOTDIR          POSIX     ++++++  Not a directory
 ENOTEMPTY        POSIX     ++++++  Directory not empty
 ENOTNAM                    +.....  Not a XENIX named type file
 ENOTRECOVERABLE  SUSv4  -  +.....  State not recoverable
 ENOTSOCK         POSIX     +++..+  Socket operation on non-socket
 ENOTSUP          POSIX     +++...  Operation not supported
 ENOTTY           POSIX     ++++++  Inappropriate I/O control operation
 ENOTUNIQ                   +.....  Name not unique on network
 ENXIO            POSIX     ++++++  No such device or address
 EOPNOTSUPP       POSIX     +++..+  Operation not supported (on socket)
 EOVERFLOW        POSIX     +++..+  Value too large to be stored in data type
 EOWNERDEAD       SUSv4     +.....  Owner died
 EPERM            POSIX  -  ++++++  Operation not permitted
 EPFNOSUPPORT               +++..+  Protocol family not supported
 EPIPE            POSIX  -  ++++++  Broken pipe
 EPROCLIM                   .++...  Too many processes
 EPROCUNAVAIL               .++...  Bad procedure for program
 EPROGMISMATCH              .++...  Program version wrong
 EPROGUNAVAIL               .++...  RPC prog. not avail
 EPROTO           POSIX     ++4...  Protocol error
 EPROTONOSUPPORT  POSIX  -  +++ss+  Protocol not supported
 EPROTOTYPE       POSIX     +++..+  Protocol wrong type for socket
 EPWROFF                    ..+...  Device power is off
 ERANGE           C89    -  ++++++  Result too large
 EREMCHG                    +.....  Remote address changed
 EREMOTE                    +++...  Object is remote
 EREMOTEIO                  +.....  Remote I/O error
 ERESTART                   +.....  Interrupted system call should be restarted
 ERFKILL                    +.....  Operation not possible due to RF-kill
 EROFS            POSIX     ++++++  Read-only file system
 ERPCMISMATCH               .++...  RPC version wrong
 ESHLIBVERS                 ..+...  Shared library version mismatch
 ESHUTDOWN                  +++..+  Cannot send after socket shutdown
 ESOCKTNOSUPPORT            +++...  Socket type not supported
 ESPIPE           POSIX     ++++++  Illegal seek
 ESRCH            POSIX     ++++++  No such process
 ESRMNT                     +.....  Srmount error
 ESTALE           POSIX     +++..+  Stale NFS file handle
 ESTRPIPE                   +.....  Streams pipe error
 ETIME            XSR       +N4...  Stream ioctl timeout
 ETIMEDOUT        POSIX  -  +++ss+  Connection timed out
 ETOOMANYREFS               +++...  Too many references: cannot splice
 ETXTBSY          POSIX     +++...  Text file busy
 EUCLEAN                    +.....  Structure needs cleaning
 EUNATCH                    +.....  Protocol driver not attached
 EUSERS                     +++...  Too many users
 EWOULDBLOCK      POSIX     +++..+  Operation would block
 EXDEV            POSIX     ++++++  Cross-device link
 EXFULL                     +.....  Exchange full
 Notations:
 F: used in FFmpeg (-: a few times, +: a lot)
 SUSv3: Single Unix Specification, version 3
 SUSv4: Single Unix Specification, version 4
 XSR: XSI STREAMS (obsolete)
 OS: availability on some supported operating systems
 L: GNU/Linux
 B: BSD (F: FreeBSD, N: NetBSD)
 M: MacOS X
 W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h)
 w: Mingw32 (3.17) and Mingw64 (2.0.1)
 b: BeOS
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -1,45 +0,0 @@
 # use pkg-config for getting CFLAGS and LDLIBS
 FFMPEG_LIBS=    libavdevice                        \
                libavformat                        \
                libavfilter                        \
                libavcodec                         \
                libswresample                      \
                libswscale                         \
                libavutil                          \
 CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 EXAMPLES=       avio_list_dir                      \
                avio_reading                       \
                decoding_encoding                  \
                demuxing_decoding                  \
                extract_mvs                        \
                filtering_video                    \
                filtering_audio                    \
                metadata                           \
                muxing                             \
                remuxing                           \
                resampling_audio                   \
                scaling_video                      \
                transcode_aac                      \
                transcoding                        \
 OBJS=$(addsuffix .o,$(EXAMPLES))
 # the following examples make explicit use of the math library
 avcodec:           LDLIBS += -lm
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm
 .phony: all clean-test clean
 all: $(OBJS) $(EXAMPLES)
 clean-test:
 	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
 clean: clean-test
 	$(RM) $(EXAMPLES) $(OBJS)
--- a/doc/examples/README
+++ b/doc/examples/README
@@ -1,23 +0,0 @@
 FFmpeg examples README
 ----------------------
 Both following use cases rely on pkg-config and make, thus make sure
 that you have them installed and working on your system.
 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.
 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 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.
--- a/doc/examples/avio_list_dir.c
+++ b/doc/examples/avio_list_dir.c
@@ -1,120 +0,0 @@
 /*
 * Copyright (c) 2014 Lukasz Marek
 *
 * 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.
 */
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavformat/avio.h>
 static const char *type_string(int type)
 {
    switch (type) {
    case AVIO_ENTRY_DIRECTORY:
        return "<DIR>";
    case AVIO_ENTRY_FILE:
        return "<FILE>";
    case AVIO_ENTRY_BLOCK_DEVICE:
        return "<BLOCK DEVICE>";
    case AVIO_ENTRY_CHARACTER_DEVICE:
        return "<CHARACTER DEVICE>";
    case AVIO_ENTRY_NAMED_PIPE:
        return "<PIPE>";
    case AVIO_ENTRY_SYMBOLIC_LINK:
        return "<LINK>";
    case AVIO_ENTRY_SOCKET:
        return "<SOCKET>";
    case AVIO_ENTRY_SERVER:
        return "<SERVER>";
    case AVIO_ENTRY_SHARE:
        return "<SHARE>";
    case AVIO_ENTRY_WORKGROUP:
        return "<WORKGROUP>";
    case AVIO_ENTRY_UNKNOWN:
    default:
        break;
    }
    return "<UNKNOWN>";
 }
 int main(int argc, char *argv[])
 {
    const char *input_dir = NULL;
    AVIODirEntry *entry = NULL;
    AVIODirContext *ctx = NULL;
    int cnt, ret;
    char filemode[4], uid_and_gid[20];
    av_log_set_level(AV_LOG_DEBUG);
    if (argc != 2) {
        fprintf(stderr, "usage: %s input_dir\n"
                "API example program to show how to list files in directory "
                "accessed through AVIOContext.\n", argv[0]);
        return 1;
    }
    input_dir = argv[1];
    /* register codecs and formats and other lavf/lavc components*/
    av_register_all();
    avformat_network_init();
    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
        goto fail;
    }
    cnt = 0;
    for (;;) {
        if ((ret = avio_read_dir(ctx, &entry)) < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
            goto fail;
        }
        if (!entry)
            break;
        if (entry->filemode == -1) {
            snprintf(filemode, 4, "???");
        } else {
            snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
        }
        snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
        if (cnt == 0)
            av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
                   "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
                   "ACCESSED", "STATUS_CHANGED");
        av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
               type_string(entry->type),
               entry->size,
               entry->name,
               uid_and_gid,
               filemode,
               entry->modification_timestamp,
               entry->access_timestamp,
               entry->status_change_timestamp);
        avio_free_directory_entry(&entry);
        cnt++;
    };
  fail:
    avio_close_dir(&ctx);
    avformat_network_deinit();
    return ret < 0 ? 1 : 0;
 }
--- a/doc/examples/avio_reading.c
+++ b/doc/examples/avio_reading.c
@@ -1,134 +0,0 @@
 /*
 * 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;
 }
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -1,665 +0,0 @@
 /*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * libavcodec API use example.
 *
 * @example decoding_encoding.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
 */
 #include <math.h>
 #include <libavutil/opt.h>
 #include <libavcodec/avcodec.h>
 #include <libavutil/channel_layout.h>
 #include <libavutil/common.h>
 #include <libavutil/imgutils.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/samplefmt.h>
 #define INBUF_SIZE 4096
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 /* check that a given sample format is supported by the encoder */
 static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
 {
    const enum AVSampleFormat *p = codec->sample_fmts;
    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
 }
 /* just pick the highest supported samplerate */
 static int select_sample_rate(AVCodec *codec)
 {
    const int *p;
    int best_samplerate = 0;
    if (!codec->supported_samplerates)
        return 44100;
    p = codec->supported_samplerates;
    while (*p) {
        best_samplerate = FFMAX(*p, best_samplerate);
        p++;
    }
    return best_samplerate;
 }
 /* select layout with the highest channel count */
 static int select_channel_layout(AVCodec *codec)
 {
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
    int best_nb_channels   = 0;
    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;
    p = codec->channel_layouts;
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);
        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return best_ch_layout;
 }
 /*
 * Audio encoding example
 */
 static void audio_encode_example(const char *filename)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket pkt;
    int i, j, k, ret, got_output;
    int buffer_size;
    FILE *f;
    uint16_t *samples;
    float t, tincr;
    printf("Encode audio file %s\n", filename);
    /* find the MP2 encoder */
    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* put sample parameters */
    c->bit_rate = 64000;
    /* check that the encoder supports s16 pcm input */
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt)) {
        fprintf(stderr, "Encoder does not support sample format %s",
                av_get_sample_fmt_name(c->sample_fmt));
        exit(1);
    }
    /* select other audio parameters supported by the encoder */
    c->sample_rate    = select_sample_rate(codec);
    c->channel_layout = select_channel_layout(codec);
    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    /* frame containing input raw audio */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }
    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
    frame->channel_layout = c->channel_layout;
    /* the codec gives us the frame size, in samples,
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
    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",
                buffer_size);
        exit(1);
    }
    /* setup the data pointers in the AVFrame */
    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                   (const uint8_t*)samples, buffer_size, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not setup audio frame\n");
        exit(1);
    }
    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for (i = 0; i < 200; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;
        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);
            for (k = 1; k < c->channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }
        /* encode the samples */
        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    /* get the delayed frames */
    for (got_output = 1; got_output; i++) {
        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    fclose(f);
    av_freep(&samples);
    av_frame_free(&frame);
    avcodec_close(c);
    av_free(c);
 }
 /*
 * Audio decoding.
 */
 static void audio_decode_example(const char *outfilename, const char *filename)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;
    av_init_packet(&avpkt);
    printf("Decode audio file %s to %s\n", filename, outfilename);
    /* find the mpeg audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }
    /* decode until eof */
    avpkt.data = inbuf;
    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
    while (avpkt.size > 0) {
        int i, ch;
        int got_frame = 0;
        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
        }
        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
            fprintf(stderr, "Error while decoding\n");
            exit(1);
        }
        if (got_frame) {
            /* if a frame has been decoded, output it */
            int data_size = av_get_bytes_per_sample(c->sample_fmt);
            if (data_size < 0) {
                /* This should not occur, checking just for paranoia */
                fprintf(stderr, "Failed to calculate data size\n");
                exit(1);
            }
            for (i=0; i<decoded_frame->nb_samples; i++)
                for (ch=0; ch<c->channels; ch++)
                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
        }
        avpkt.size -= len;
        avpkt.data += len;
        avpkt.dts =
        avpkt.pts = AV_NOPTS_VALUE;
        if (avpkt.size < AUDIO_REFILL_THRESH) {
            /* Refill the input buffer, to avoid trying to decode
             * incomplete frames. Instead of this, one could also use
             * a parser, or use a proper container format through
             * libavformat. */
            memmove(inbuf, avpkt.data, avpkt.size);
            avpkt.data = inbuf;
            len = fread(avpkt.data + avpkt.size, 1,
                        AUDIO_INBUF_SIZE - avpkt.size, f);
            if (len > 0)
                avpkt.size += len;
        }
    }
    fclose(outfile);
    fclose(f);
    avcodec_close(c);
    av_free(c);
    av_frame_free(&decoded_frame);
 }
 /*
 * Video encoding example
 */
 static void video_encode_example(const char *filename, int codec_id)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int i, ret, x, y, got_output;
    FILE *f;
    AVFrame *frame;
    AVPacket pkt;
    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
    printf("Encode video file %s\n", filename);
    /* find the mpeg1 video encoder */
    codec = avcodec_find_encoder(codec_id);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate video codec context\n");
        exit(1);
    }
    /* put sample parameters */
    c->bit_rate = 400000;
    /* resolution must be a multiple of two */
    c->width = 352;
    c->height = 288;
    /* frames per second */
    c->time_base = (AVRational){1,25};
    /* emit one intra frame every ten frames
     * check frame pict_type before passing frame
     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
     * then gop_size is ignored and the output of encoder
     * will always be I frame irrespective to gop_size
     */
    c->gop_size = 10;
    c->max_b_frames = 1;
    c->pix_fmt = AV_PIX_FMT_YUV420P;
    if (codec_id == AV_CODEC_ID_H264)
        av_opt_set(c->priv_data, "preset", "slow", 0);
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    frame = 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;
    /* the image can be allocated by any means and av_image_alloc() is
     * just the most convenient way if av_malloc() is to be used */
    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
                         c->pix_fmt, 32);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate raw picture buffer\n");
        exit(1);
    }
    /* encode 1 second of video */
    for (i = 0; i < 25; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL;    // packet data will be allocated by the encoder
        pkt.size = 0;
        fflush(stdout);
        /* prepare a dummy image */
        /* Y */
        for (y = 0; y < c->height; y++) {
            for (x = 0; x < c->width; x++) {
                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
            }
        }
        /* Cb and Cr */
        for (y = 0; y < c->height/2; y++) {
            for (x = 0; x < c->width/2; x++) {
                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
            }
        }
        frame->pts = i;
        /* encode the image */
        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    /* get the delayed frames */
    for (got_output = 1; got_output; i++) {
        fflush(stdout);
        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    /* add sequence end code to have a real mpeg file */
    fwrite(endcode, 1, sizeof(endcode), f);
    fclose(f);
    avcodec_close(c);
    av_free(c);
    av_freep(&frame->data[0]);
    av_frame_free(&frame);
    printf("\n");
 }
 /*
 * Video decoding example
 */
 static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
                     char *filename)
 {
    FILE *f;
    int i;
    f = fopen(filename,"w");
    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
    for (i = 0; i < ysize; i++)
        fwrite(buf + i * wrap, 1, xsize, f);
    fclose(f);
 }
 static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
 {
    int len, got_frame;
    char buf[1024];
    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
    if (len < 0) {
        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
        return len;
    }
    if (got_frame) {
        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
        fflush(stdout);
        /* the picture is allocated by the decoder, no need to free it */
        snprintf(buf, sizeof(buf), outfilename, *frame_count);
        pgm_save(frame->data[0], frame->linesize[0],
                 avctx->width, avctx->height, buf);
        (*frame_count)++;
    }
    if (pkt->data) {
        pkt->size -= len;
        pkt->data += len;
    }
    return 0;
 }
 static void video_decode_example(const char *outfilename, const char *filename)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int frame_count;
    FILE *f;
    AVFrame *frame;
    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    av_init_packet(&avpkt);
    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
    printf("Decode video file %s to %s\n", filename, outfilename);
    /* find the mpeg1 video decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate video codec context\n");
        exit(1);
    }
    if(codec->capabilities&CODEC_CAP_TRUNCATED)
        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
    /* For some codecs, such as msmpeg4 and mpeg4, width and height
       MUST be initialized there because this information is not
       available in the bitstream. */
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
    frame_count = 0;
    for (;;) {
        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
        if (avpkt.size == 0)
            break;
        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
           and this is the only method to use them because you cannot
           know the compressed data size before analysing it.
           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
           based, so you must call them with all the data for one
           frame exactly. You must also initialize 'width' and
           'height' before initializing them. */
        /* NOTE2: some codecs allow the raw parameters (frame size,
           sample rate) to be changed at any frame. We handle this, so
           you should also take care of it */
        /* here, we use a stream based decoder (mpeg1video), so we
           feed decoder and see if it could decode a frame */
        avpkt.data = inbuf;
        while (avpkt.size > 0)
            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
                exit(1);
    }
    /* some codecs, such as MPEG, transmit the I and P frame with a
       latency of one frame. You must do the following to have a
       chance to get the last frame of the video */
    avpkt.data = NULL;
    avpkt.size = 0;
    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
    fclose(f);
    avcodec_close(c);
    av_free(c);
    av_frame_free(&frame);
    printf("\n");
 }
 int main(int argc, char **argv)
 {
    const char *output_type;
    /* register all the codecs */
    avcodec_register_all();
    if (argc < 2) {
        printf("usage: %s output_type\n"
               "API example program to decode/encode a media stream with libavcodec.\n"
               "This program generates a synthetic stream and encodes it to a file\n"
               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
               "The encoded stream is then decoded and written to a raw data output.\n"
               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
               argv[0]);
        return 1;
    }
    output_type = argv[1];
    if (!strcmp(output_type, "h264")) {
        video_encode_example("test.h264", AV_CODEC_ID_H264);
    } else if (!strcmp(output_type, "mp2")) {
        audio_encode_example("test.mp2");
        audio_decode_example("test.pcm", "test.mp2");
    } else if (!strcmp(output_type, "mpg")) {
        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
        video_decode_example("test%02d.pgm", "test.mpg");
    } else {
        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
                output_type);
        return 1;
    }
    return 0;
 }
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@@ -1,407 +0,0 @@
 /*
 * Copyright (c) 2012 Stefano Sabatini
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * Demuxing and decoding example.
 *
 * Show how to use the libavformat and libavcodec API to demux and
 * decode audio and video data.
 * @example demuxing_decoding.c
 */
 #include <libavutil/imgutils.h>
 #include <libavutil/samplefmt.h>
 #include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 static AVFormatContext *fmt_ctx = NULL;
 static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
 static int width, height;
 static enum AVPixelFormat pix_fmt;
 static AVStream *video_stream = NULL, *audio_stream = NULL;
 static const char *src_filename = NULL;
 static const char *video_dst_filename = NULL;
 static const char *audio_dst_filename = NULL;
 static FILE *video_dst_file = NULL;
 static FILE *audio_dst_file = NULL;
 static uint8_t *video_dst_data[4] = {NULL};
 static int      video_dst_linesize[4];
 static int video_dst_bufsize;
 static 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 (%s)\n", av_err2str(ret));
            return ret;
        }
        if (*got_frame) {
            if (frame->width != width || frame->height != height ||
                frame->format != pix_fmt) {
                /* To handle this change, one could call av_image_alloc again and
                 * decode the following frames into another rawvideo file. */
                fprintf(stderr, "Error: Width, height and pixel format have to be "
                        "constant in a rawvideo file, but the width, height or "
                        "pixel format of the input video changed:\n"
                        "old: width = %d, height = %d, format = %s\n"
                        "new: width = %d, height = %d, format = %s\n",
                        width, height, av_get_pix_fmt_name(pix_fmt),
                        frame->width, frame->height,
                        av_get_pix_fmt_name(frame->format));
                return -1;
            }
            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   video_frame_count++, frame->coded_picture_number,
                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
            /* copy decoded frame to destination buffer:
             * this is required since rawvideo expects non aligned data */
            av_image_copy(video_dst_data, video_dst_linesize,
                          (const uint8_t **)(frame->data), frame->linesize,
                          pix_fmt, width, height);
            /* write to rawvideo file */
            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
        }
    } else if (pkt.stream_index == audio_stream_idx) {
        /* decode audio frame */
        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
            fprintf(stderr, "Error decoding audio frame (%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));
            /* 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);
        }
    }
    /* 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,
                              AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
    int ret, stream_index;
    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) {
        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
                av_get_media_type_string(type), src_filename);
        return ret;
    } else {
        stream_index = ret;
        st = fmt_ctx->streams[stream_index];
        /* find decoder for the stream */
        dec_ctx = st->codec;
        dec = avcodec_find_decoder(dec_ctx->codec_id);
        if (!dec) {
            fprintf(stderr, "Failed to find %s codec\n",
                    av_get_media_type_string(type));
            return AVERROR(EINVAL);
        }
        /* 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;
        }
        *stream_idx = stream_index;
    }
    return 0;
 }
 static int get_format_from_sample_fmt(const char **fmt,
                                      enum AVSampleFormat sample_fmt)
 {
    int i;
    struct sample_fmt_entry {
        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
    } sample_fmt_entries[] = {
        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
    };
    *fmt = NULL;
    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
        if (sample_fmt == entry->sample_fmt) {
            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
            return 0;
        }
    }
    fprintf(stderr,
            "sample format %s is not supported as output format\n",
            av_get_sample_fmt_name(sample_fmt));
    return -1;
 }
 int main (int argc, char **argv)
 {
    int ret = 0, got_frame;
    if (argc != 4 && 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\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];
    /* register all formats and codecs */
    av_register_all();
    /* open input file, and allocate format context */
    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
        fprintf(stderr, "Could not open source file %s\n", src_filename);
        exit(1);
    }
    /* retrieve stream information */
    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
        fprintf(stderr, "Could not find stream information\n");
        exit(1);
    }
    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
        video_stream = fmt_ctx->streams[video_stream_idx];
        video_dec_ctx = video_stream->codec;
        video_dst_file = fopen(video_dst_filename, "wb");
        if (!video_dst_file) {
            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
            ret = 1;
            goto end;
        }
        /* allocate image where the decoded image will be put */
        width = video_dec_ctx->width;
        height = video_dec_ctx->height;
        pix_fmt = video_dec_ctx->pix_fmt;
        ret = av_image_alloc(video_dst_data, video_dst_linesize,
                             width, height, pix_fmt, 1);
        if (ret < 0) {
            fprintf(stderr, "Could not allocate raw video buffer\n");
            goto end;
        }
        video_dst_bufsize = ret;
    }
    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
        audio_stream = fmt_ctx->streams[audio_stream_idx];
        audio_dec_ctx = audio_stream->codec;
        audio_dst_file = fopen(audio_dst_filename, "wb");
        if (!audio_dst_file) {
            fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
            ret = 1;
            goto end;
        }
    }
    /* dump input information to stderr */
    av_dump_format(fmt_ctx, 0, src_filename, 0);
    if (!audio_stream && !video_stream) {
        fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
        ret = 1;
        goto end;
    }
    /* 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);
        goto end;
    }
    /* initialize packet, set data to NULL, let the demuxer fill it */
    av_init_packet(&pkt);
    pkt.data = NULL;
    pkt.size = 0;
    if (video_stream)
        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
    if (audio_stream)
        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
    /* read frames from the file */
    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
        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 */
    pkt.data = NULL;
    pkt.size = 0;
    do {
        decode_packet(&got_frame, 1);
    } while (got_frame);
    printf("Demuxing succeeded.\n");
    if (video_stream) {
        printf("Play the output video file with the command:\n"
               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
               av_get_pix_fmt_name(pix_fmt), width, height,
               video_dst_filename);
    }
    if (audio_stream) {
        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
        int n_channels = audio_dec_ctx->channels;
        const char *fmt;
        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, n_channels, audio_dec_ctx->sample_rate,
               audio_dst_filename);
    }
 end:
    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);
    if (api_mode == API_MODE_OLD)
        avcodec_free_frame(&frame);
    else
        av_frame_free(&frame);
    av_free(video_dst_data[0]);
    return ret < 0;
 }
--- a/doc/examples/extract_mvs.c
+++ b/doc/examples/extract_mvs.c
@@ -1,185 +0,0 @@
 /*
 * Copyright (c) 2012 Stefano Sabatini
 * Copyright (c) 2014 Clément Bœsch
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 #include <libavutil/motion_vector.h>
 #include <libavformat/avformat.h>
 static AVFormatContext *fmt_ctx = NULL;
 static AVCodecContext *video_dec_ctx = NULL;
 static AVStream *video_stream = NULL;
 static const char *src_filename = NULL;
 static int video_stream_idx = -1;
 static AVFrame *frame = NULL;
 static AVPacket pkt;
 static int video_frame_count = 0;
 static int decode_packet(int *got_frame, int cached)
 {
    int decoded = pkt.size;
    *got_frame = 0;
    if (pkt.stream_index == video_stream_idx) {
        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
            return ret;
        }
        if (*got_frame) {
            int i;
            AVFrameSideData *sd;
            video_frame_count++;
            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
            if (sd) {
                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
                    const AVMotionVector *mv = &mvs[i];
                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
                           video_frame_count, mv->source,
                           mv->w, mv->h, mv->src_x, mv->src_y,
                           mv->dst_x, mv->dst_y, mv->flags);
                }
            }
        }
    }
    return decoded;
 }
 static int open_codec_context(int *stream_idx,
                              AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
    int ret;
    AVStream *st;
    AVCodecContext *dec_ctx = NULL;
    AVCodec *dec = NULL;
    AVDictionary *opts = NULL;
    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
                av_get_media_type_string(type), src_filename);
        return ret;
    } else {
        *stream_idx = ret;
        st = fmt_ctx->streams[*stream_idx];
        /* find decoder for the stream */
        dec_ctx = st->codec;
        dec = avcodec_find_decoder(dec_ctx->codec_id);
        if (!dec) {
            fprintf(stderr, "Failed to find %s codec\n",
                    av_get_media_type_string(type));
            return AVERROR(EINVAL);
        }
        /* Init the video decoder */
        av_dict_set(&opts, "flags2", "+export_mvs", 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;
        }
    }
    return 0;
 }
 int main(int argc, char **argv)
 {
    int ret = 0, got_frame;
    if (argc != 2) {
        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
        exit(1);
    }
    src_filename = argv[1];
    av_register_all();
    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
        fprintf(stderr, "Could not open source file %s\n", src_filename);
        exit(1);
    }
    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
        fprintf(stderr, "Could not find stream information\n");
        exit(1);
    }
    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
        video_stream = fmt_ctx->streams[video_stream_idx];
        video_dec_ctx = video_stream->codec;
    }
    av_dump_format(fmt_ctx, 0, src_filename, 0);
    if (!video_stream) {
        fprintf(stderr, "Could not find video stream in the input, aborting\n");
        ret = 1;
        goto end;
    }
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }
    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
    /* initialize packet, set data to NULL, let the demuxer fill it */
    av_init_packet(&pkt);
    pkt.data = NULL;
    pkt.size = 0;
    /* read frames from the file */
    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
        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 */
    pkt.data = NULL;
    pkt.size = 0;
    do {
        decode_packet(&got_frame, 1);
    } while (got_frame);
 end:
    avcodec_close(video_dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    return ret < 0;
 }
--- a/doc/examples/filter_audio.c
+++ b/doc/examples/filter_audio.c
@@ -1,365 +0,0 @@
 /*
 * 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/mem.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;
 }
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -1,296 +0,0 @@
 /*
 * Copyright (c) 2010 Nicolas George
 * Copyright (c) 2011 Stefano Sabatini
 * Copyright (c) 2012 Clément Bœsch
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * API example for audio decoding and filtering
 * @example filtering_audio.c
 */
 #include <unistd.h>
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
 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;
 AVFilterContext *buffersink_ctx;
 AVFilterContext *buffersrc_ctx;
 AVFilterGraph *filter_graph;
 static int audio_stream_index = -1;
 static int open_input_file(const char *filename)
 {
    int ret;
    AVCodec *dec;
    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
        return ret;
    }
    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
        return ret;
    }
    /* select the audio stream */
    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
        return ret;
    }
    audio_stream_index = ret;
    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
        return ret;
    }
    return 0;
 }
 static int init_filters(const char *filters_descr)
 {
    char args[512];
    int ret = 0;
    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
    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)
        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
    snprintf(args, sizeof(args),
            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
             time_base.num, time_base.den, dec_ctx->sample_rate,
             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
        goto end;
    }
    /* buffer audio sink: to terminate the filter chain. */
    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
        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;
    }
    /*
     * Set the endpoints for the filter graph. The filter_graph will
     * be linked to the graph described by filters_descr.
     */
    /*
     * The buffer source output must be connected to the input pad of
     * the first filter described by filters_descr; since the first
     * filter input label is not specified, it is set to "in" by
     * default.
     */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
    outputs->pad_idx    = 0;
    outputs->next       = NULL;
    /*
     * The buffer sink input must be connected to the output pad of
     * the last filter described by filters_descr; since the last
     * filter output label is not specified, it is set to "out" by
     * default.
     */
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
    inputs->next       = NULL;
    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)
        goto end;
    /* Print summary of the sink buffer
     * Note: args buffer is reused to store channel layout string */
    outlink = buffersink_ctx->inputs[0];
    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
           (int)outlink->sample_rate,
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
           args);
 end:
    avfilter_inout_free(&inputs);
    avfilter_inout_free(&outputs);
    return ret;
 }
 static void print_frame(const AVFrame *frame)
 {
    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) {
        fputc(*p    & 0xff, stdout);
        fputc(*p>>8 & 0xff, stdout);
        p++;
    }
    fflush(stdout);
 }
 int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet0, packet;
    AVFrame *frame = av_frame_alloc();
    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;
    if (!frame || !filt_frame) {
        perror("Could not allocate frame");
        exit(1);
    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
        exit(1);
    }
    av_register_all();
    avfilter_register_all();
    if ((ret = open_input_file(argv[1])) < 0)
        goto end;
    if ((ret = init_filters(filter_descr)) < 0)
        goto end;
    /* read all packets */
    packet0.data = NULL;
    packet.data = NULL;
    while (1) {
        if (!packet0.data) {
            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
                break;
            packet0 = packet;
        }
        if (packet.stream_index == audio_stream_index) {
            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_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_frame(buffersink_ctx, filt_frame);
                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
                    if (ret < 0)
                        goto end;
                    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);
        }
    }
 end:
    avfilter_graph_free(&filter_graph);
    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);
    if (ret < 0 && ret != AVERROR_EOF) {
        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }
    exit(0);
 }
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -1,278 +0,0 @@
 /*
 * Copyright (c) 2010 Nicolas George
 * Copyright (c) 2011 Stefano Sabatini
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * API example for decoding and filtering
 * @example filtering_video.c
 */
 #define _XOPEN_SOURCE 600 /* for usleep */
 #include <unistd.h>
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
 const char *filter_descr = "scale=78:24";
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
 AVFilterContext *buffersink_ctx;
 AVFilterContext *buffersrc_ctx;
 AVFilterGraph *filter_graph;
 static int video_stream_index = -1;
 static int64_t last_pts = AV_NOPTS_VALUE;
 static int open_input_file(const char *filename)
 {
    int ret;
    AVCodec *dec;
    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
        return ret;
    }
    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
        return ret;
    }
    /* select the video stream */
    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
        return ret;
    }
    video_stream_index = ret;
    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
    /* init the video decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
        return ret;
    }
    return 0;
 }
 static int init_filters(const char *filters_descr)
 {
    char args[512];
    int ret = 0;
    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
    AVFilter *buffersink = avfilter_get_by_name("buffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
    AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
    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),
            "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
            dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
            time_base.num, time_base.den,
            dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
    ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
        goto end;
    }
    /* buffer video sink: to terminate the filter chain. */
    ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
        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;
    }
    /*
     * Set the endpoints for the filter graph. The filter_graph will
     * be linked to the graph described by filters_descr.
     */
    /*
     * The buffer source output must be connected to the input pad of
     * the first filter described by filters_descr; since the first
     * filter input label is not specified, it is set to "in" by
     * default.
     */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
    outputs->pad_idx    = 0;
    outputs->next       = NULL;
    /*
     * The buffer sink input must be connected to the output pad of
     * the last filter described by filters_descr; since the last
     * filter output label is not specified, it is set to "out" by
     * default.
     */
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
    inputs->next       = NULL;
    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)
        goto end;
 end:
    avfilter_inout_free(&inputs);
    avfilter_inout_free(&outputs);
    return ret;
 }
 static void display_frame(const AVFrame *frame, AVRational time_base)
 {
    int x, y;
    uint8_t *p0, *p;
    int64_t delay;
    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(frame->pts - last_pts,
                                 time_base, AV_TIME_BASE_Q);
            if (delay > 0 && delay < 1000000)
                usleep(delay);
        }
        last_pts = frame->pts;
    }
    /* Trivial ASCII grayscale display. */
    p0 = frame->data[0];
    puts("\033c");
    for (y = 0; y < frame->height; y++) {
        p = p0;
        for (x = 0; x < frame->width; x++)
            putchar(" .-+#"[*(p++) / 52]);
        putchar('\n');
        p0 += frame->linesize[0];
    }
    fflush(stdout);
 }
 int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
    AVFrame *frame = av_frame_alloc();
    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;
    if (!frame || !filt_frame) {
        perror("Could not allocate frame");
        exit(1);
    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file\n", argv[0]);
        exit(1);
    }
    av_register_all();
    avfilter_register_all();
    if ((ret = open_input_file(argv[1])) < 0)
        goto end;
    if ((ret = init_filters(filter_descr)) < 0)
        goto end;
    /* read all packets */
    while (1) {
        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;
        if (packet.stream_index == video_stream_index) {
            got_frame = 0;
            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                break;
            }
            if (got_frame) {
                frame->pts = av_frame_get_best_effort_timestamp(frame);
                /* push the decoded frame into the filtergraph */
                if (av_buffersrc_add_frame_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 frames from the filtergraph */
                while (1) {
                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
                    if (ret < 0)
                        goto end;
                    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);
    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);
    if (ret < 0 && ret != AVERROR_EOF) {
        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }
    exit(0);
 }
--- a/doc/examples/metadata.c
+++ b/doc/examples/metadata.c
@@ -1,56 +0,0 @@
 /*
 * Copyright (c) 2011 Reinhard Tartler
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * Shows how the metadata API can be used in application programs.
 * @example metadata.c
 */
 #include <stdio.h>
 #include <libavformat/avformat.h>
 #include <libavutil/dict.h>
 int main (int argc, char **argv)
 {
    AVFormatContext *fmt_ctx = NULL;
    AVDictionaryEntry *tag = NULL;
    int ret;
    if (argc != 2) {
        printf("usage: %s <input_file>\n"
               "example program to demonstrate the use of the libavformat metadata API.\n"
               "\n", argv[0]);
        return 1;
    }
    av_register_all();
    if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
        return ret;
    while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
        printf("%s=%s\n", tag->key, tag->value);
    avformat_close_input(&fmt_ctx);
    return 0;
 }
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -1,670 +0,0 @@
 /*
 * Copyright (c) 2003 Fabrice Bellard
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * libavformat API example.
 *
 * Output a media file in any supported libavformat format. The default
 * codecs are used.
 * @example muxing.c
 */
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
 #include <math.h>
 #include <libavutil/avassert.h>
 #include <libavutil/channel_layout.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>
 #define STREAM_DURATION   10.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
 #define SCALE_FLAGS SWS_BICUBIC
 // a wrapper around a single output AVStream
 typedef struct OutputStream {
    AVStream *st;
    /* pts of the next frame that will be generated */
    int64_t next_pts;
    int samples_count;
    AVFrame *frame;
    AVFrame *tmp_frame;
    float t, tincr, tincr2;
    struct SwsContext *sws_ctx;
    struct SwrContext *swr_ctx;
 } OutputStream;
 static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
 {
    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
    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 */
    av_packet_rescale_ts(pkt, *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 void add_stream(OutputStream *ost, AVFormatContext *oc,
                       AVCodec **codec,
                       enum AVCodecID codec_id)
 {
    AVCodecContext *c;
    int i;
    /* find the encoder */
    *codec = avcodec_find_encoder(codec_id);
    if (!(*codec)) {
        fprintf(stderr, "Could not find encoder for '%s'\n",
                avcodec_get_name(codec_id));
        exit(1);
    }
    ost->st = avformat_new_stream(oc, *codec);
    if (!ost->st) {
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }
    ost->st->id = oc->nb_streams-1;
    c = ost->st->codec;
    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
        c->sample_fmt  = (*codec)->sample_fmts ?
            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
        if ((*codec)->supported_samplerates) {
            c->sample_rate = (*codec)->supported_samplerates[0];
            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
                if ((*codec)->supported_samplerates[i] == 44100)
                    c->sample_rate = 44100;
            }
        }
        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
        c->channel_layout = AV_CH_LAYOUT_STEREO;
        if ((*codec)->channel_layouts) {
            c->channel_layout = (*codec)->channel_layouts[0];
            for (i = 0; (*codec)->channel_layouts[i]; i++) {
                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
                    c->channel_layout = AV_CH_LAYOUT_STEREO;
            }
        }
        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
        ost->st->time_base = (AVRational){ 1, c->sample_rate };
        break;
    case AVMEDIA_TYPE_VIDEO:
        c->codec_id = codec_id;
        c->bit_rate = 400000;
        /* Resolution must be a multiple of two. */
        c->width    = 352;
        c->height   = 288;
        /* timebase: This is the fundamental unit of time (in seconds) in terms
         * of which frame timestamps are represented. For fixed-fps content,
         * timebase should be 1/framerate and timestamp increments should be
         * identical to 1. */
        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
        c->time_base       = ost->st->time_base;
        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
        c->pix_fmt       = STREAM_PIX_FMT;
        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
            /* just for testing, we also add B frames */
            c->max_b_frames = 2;
        }
        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
            /* Needed to avoid using macroblocks in which some coeffs overflow.
             * This does not happen with normal video, it just happens here as
             * the motion of the chroma plane does not match the luma plane. */
            c->mb_decision = 2;
        }
    break;
    default:
        break;
    }
    /* Some formats want stream headers to be separate. */
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
 }
 /**************************************************************/
 /* audio output */
 static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
                                  uint64_t channel_layout,
                                  int sample_rate, int nb_samples)
 {
    AVFrame *frame = av_frame_alloc();
    int ret;
    if (!frame) {
        fprintf(stderr, "Error allocating an audio frame\n");
        exit(1);
    }
    frame->format = sample_fmt;
    frame->channel_layout = channel_layout;
    frame->sample_rate = sample_rate;
    frame->nb_samples = nb_samples;
    if (nb_samples) {
        ret = av_frame_get_buffer(frame, 0);
        if (ret < 0) {
            fprintf(stderr, "Error allocating an audio buffer\n");
            exit(1);
        }
    }
    return frame;
 }
 static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
    AVCodecContext *c;
    int nb_samples;
    int ret;
    AVDictionary *opt = NULL;
    c = ost->st->codec;
    /* open it */
    av_dict_copy(&opt, opt_arg, 0);
    ret = avcodec_open2(c, codec, &opt);
    av_dict_free(&opt);
    if (ret < 0) {
        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
        exit(1);
    }
    /* init signal generator */
    ost->t     = 0;
    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
    /* increment frequency by 110 Hz per second */
    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
        nb_samples = 10000;
    else
        nb_samples = c->frame_size;
    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
                                       c->sample_rate, nb_samples);
    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
                                       c->sample_rate, nb_samples);
    /* create resampler context */
        ost->swr_ctx = swr_alloc();
        if (!ost->swr_ctx) {
            fprintf(stderr, "Could not allocate resampler context\n");
            exit(1);
        }
        /* set options */
        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
        /* initialize the resampling context */
        if ((ret = swr_init(ost->swr_ctx)) < 0) {
            fprintf(stderr, "Failed to initialize the resampling context\n");
            exit(1);
        }
 }
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
 * 'nb_channels' channels. */
 static AVFrame *get_audio_frame(OutputStream *ost)
 {
    AVFrame *frame = ost->tmp_frame;
    int j, i, v;
    int16_t *q = (int16_t*)frame->data[0];
    /* check if we want to generate more frames */
    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
        return NULL;
    for (j = 0; j <frame->nb_samples; j++) {
        v = (int)(sin(ost->t) * 10000);
        for (i = 0; i < ost->st->codec->channels; i++)
            *q++ = v;
        ost->t     += ost->tincr;
        ost->tincr += ost->tincr2;
    }
    frame->pts = ost->next_pts;
    ost->next_pts  += frame->nb_samples;
    return frame;
 }
 /*
 * encode one audio frame and send it to the muxer
 * return 1 when encoding is finished, 0 otherwise
 */
 static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
 {
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
    AVFrame *frame;
    int ret;
    int got_packet;
    int dst_nb_samples;
    av_init_packet(&pkt);
    c = ost->st->codec;
    frame = get_audio_frame(ost);
    if (frame) {
        /* convert samples from native format to destination codec format, using the resampler */
            /* compute destination number of samples */
            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
            av_assert0(dst_nb_samples == frame->nb_samples);
        /* when we pass a frame to the encoder, it may keep a reference to it
         * internally;
         * make sure we do not overwrite it here
         */
        ret = av_frame_make_writable(ost->frame);
        if (ret < 0)
            exit(1);
            /* convert to destination format */
            ret = swr_convert(ost->swr_ctx,
                              ost->frame->data, dst_nb_samples,
                              (const uint8_t **)frame->data, frame->nb_samples);
            if (ret < 0) {
                fprintf(stderr, "Error while converting\n");
                exit(1);
            }
            frame = ost->frame;
        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
        ost->samples_count += dst_nb_samples;
    }
    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
    if (ret < 0) {
        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
        exit(1);
    }
    if (got_packet) {
        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
        if (ret < 0) {
            fprintf(stderr, "Error while writing audio frame: %s\n",
                    av_err2str(ret));
            exit(1);
        }
    }
    return (frame || got_packet) ? 0 : 1;
 }
 /**************************************************************/
 /* video output */
 static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
 {
    AVFrame *picture;
    int ret;
    picture = av_frame_alloc();
    if (!picture)
        return NULL;
    picture->format = pix_fmt;
    picture->width  = width;
    picture->height = height;
    /* allocate the buffers for the frame data */
    ret = av_frame_get_buffer(picture, 32);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate frame data.\n");
        exit(1);
    }
    return picture;
 }
 static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
    int ret;
    AVCodecContext *c = ost->st->codec;
    AVDictionary *opt = NULL;
    av_dict_copy(&opt, opt_arg, 0);
    /* open the codec */
    ret = avcodec_open2(c, codec, &opt);
    av_dict_free(&opt);
    if (ret < 0) {
        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
        exit(1);
    }
    /* allocate and init a re-usable frame */
    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
    if (!ost->frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
    /* If the output format is not YUV420P, then a temporary YUV420P
     * picture is needed too. It is then converted to the required
     * output format. */
    ost->tmp_frame = NULL;
    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
        if (!ost->tmp_frame) {
            fprintf(stderr, "Could not allocate temporary picture\n");
            exit(1);
        }
    }
 }
 /* Prepare a dummy image. */
 static void fill_yuv_image(AVFrame *pict, int frame_index,
                           int width, int height)
 {
    int x, y, i, ret;
    /* when we pass a frame to the encoder, it may keep a reference to it
     * internally;
     * make sure we do not overwrite it here
     */
    ret = av_frame_make_writable(pict);
    if (ret < 0)
        exit(1);
    i = frame_index;
    /* Y */
    for (y = 0; y < height; y++)
        for (x = 0; x < width; x++)
            pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
    /* Cb and Cr */
    for (y = 0; y < height / 2; y++) {
        for (x = 0; x < width / 2; x++) {
            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
        }
    }
 }
 static AVFrame *get_video_frame(OutputStream *ost)
 {
    AVCodecContext *c = ost->st->codec;
    /* check if we want to generate more frames */
    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
        return NULL;
    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
        /* as we only generate a YUV420P picture, we must convert it
         * to the codec pixel format if needed */
        if (!ost->sws_ctx) {
            ost->sws_ctx = sws_getContext(c->width, c->height,
                                          AV_PIX_FMT_YUV420P,
                                          c->width, c->height,
                                          c->pix_fmt,
                                          SCALE_FLAGS, NULL, NULL, NULL);
            if (!ost->sws_ctx) {
                fprintf(stderr,
                        "Could not initialize the conversion context\n");
                exit(1);
            }
        }
        fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
        sws_scale(ost->sws_ctx,
                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
                  0, c->height, ost->frame->data, ost->frame->linesize);
    } else {
        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
    }
    ost->frame->pts = ost->next_pts++;
    return ost->frame;
 }
 /*
 * encode one video frame and send it to the muxer
 * return 1 when encoding is finished, 0 otherwise
 */
 static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
 {
    int ret;
    AVCodecContext *c;
    AVFrame *frame;
    int got_packet = 0;
    c = ost->st->codec;
    frame = get_video_frame(ost);
    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
        /* a hack to avoid data copy with some raw video muxers */
        AVPacket pkt;
        av_init_packet(&pkt);
        if (!frame)
            return 1;
        pkt.flags        |= AV_PKT_FLAG_KEY;
        pkt.stream_index  = ost->st->index;
        pkt.data          = (uint8_t *)frame;
        pkt.size          = sizeof(AVPicture);
        pkt.pts = pkt.dts = frame->pts;
        av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
        AVPacket pkt = { 0 };
        av_init_packet(&pkt);
        /* encode the image */
        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
        if (ret < 0) {
            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
            exit(1);
        }
        if (got_packet) {
            ret = write_frame(oc, &c->time_base, ost->st, &pkt);
        } else {
            ret = 0;
        }
    }
    if (ret < 0) {
        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
        exit(1);
    }
    return (frame || got_packet) ? 0 : 1;
 }
 static void close_stream(AVFormatContext *oc, OutputStream *ost)
 {
    avcodec_close(ost->st->codec);
    av_frame_free(&ost->frame);
    av_frame_free(&ost->tmp_frame);
    sws_freeContext(ost->sws_ctx);
    swr_free(&ost->swr_ctx);
 }
 /**************************************************************/
 /* media file output */
 int main(int argc, char **argv)
 {
    OutputStream video_st = { 0 }, audio_st = { 0 };
    const char *filename;
    AVOutputFormat *fmt;
    AVFormatContext *oc;
    AVCodec *audio_codec, *video_codec;
    int ret;
    int have_video = 0, have_audio = 0;
    int encode_video = 0, encode_audio = 0;
    AVDictionary *opt = NULL;
    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();
    if (argc < 2) {
        printf("usage: %s output_file\n"
               "API example program to output a media file with libavformat.\n"
               "This program generates a synthetic audio and video stream, encodes and\n"
               "muxes them into a file named output_file.\n"
               "The output format is automatically guessed according to the file extension.\n"
               "Raw images can also be output by using '%%d' in the filename.\n"
               "\n", argv[0]);
        return 1;
    }
    filename = argv[1];
    if (argc > 3 && !strcmp(argv[2], "-flags")) {
        av_dict_set(&opt, argv[2]+1, argv[3], 0);
    }
    /* allocate the output media context */
    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
    if (!oc) {
        printf("Could not deduce output format from file extension: using MPEG.\n");
        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
    }
    if (!oc)
        return 1;
    fmt = oc->oformat;
    /* Add the audio and video streams using the default format codecs
     * and initialize the codecs. */
    if (fmt->video_codec != AV_CODEC_ID_NONE) {
        add_stream(&video_st, oc, &video_codec, fmt->video_codec);
        have_video = 1;
        encode_video = 1;
    }
    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
        add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
        have_audio = 1;
        encode_audio = 1;
    }
    /* Now that all the parameters are set, we can open the audio and
     * video codecs and allocate the necessary encode buffers. */
    if (have_video)
        open_video(oc, video_codec, &video_st, opt);
    if (have_audio)
        open_audio(oc, audio_codec, &audio_st, opt);
    av_dump_format(oc, 0, filename, 1);
    /* open the output file, if needed */
    if (!(fmt->flags & AVFMT_NOFILE)) {
        ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
        if (ret < 0) {
            fprintf(stderr, "Could not open '%s': %s\n", filename,
                    av_err2str(ret));
            return 1;
        }
    }
    /* Write the stream header, if any. */
    ret = avformat_write_header(oc, &opt);
    if (ret < 0) {
        fprintf(stderr, "Error occurred when opening output file: %s\n",
                av_err2str(ret));
        return 1;
    }
    while (encode_video || encode_audio) {
        /* select the stream to encode */
        if (encode_video &&
            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
            encode_video = !write_video_frame(oc, &video_st);
        } else {
            encode_audio = !write_audio_frame(oc, &audio_st);
        }
    }
    /* Write the trailer, if any. The trailer must be written before you
     * close the CodecContexts open when you wrote the header; otherwise
     * av_write_trailer() may try to use memory that was freed on
     * av_codec_close(). */
    av_write_trailer(oc);
    /* Close each codec. */
    if (have_video)
        close_stream(oc, &video_st);
    if (have_audio)
        close_stream(oc, &audio_st);
    if (!(fmt->flags & AVFMT_NOFILE))
        /* Close the output file. */
        avio_closep(&oc->pb);
    /* free the stream */
    avformat_free_context(oc);
    return 0;
 }
--- a/doc/examples/qsvdec.c
+++ b/doc/examples/qsvdec.c
@@ -1,484 +0,0 @@
 /*
 * Copyright (c) 2015 Anton Khirnov
 *
 * 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
 * Intel QSV-accelerated H.264 decoding example.
 *
 * @example qsvdec.c
 * This example shows how to do QSV-accelerated H.264 decoding with output
 * frames in the VA-API video surfaces.
 */
 #include "config.h"
 #include <stdio.h>
 #include <mfx/mfxvideo.h>
 #include <va/va.h>
 #include <va/va_x11.h>
 #include <X11/Xlib.h>
 #include "libavformat/avformat.h"
 #include "libavformat/avio.h"
 #include "libavcodec/avcodec.h"
 #include "libavcodec/qsv.h"
 #include "libavutil/error.h"
 #include "libavutil/mem.h"
 typedef struct DecodeContext {
    mfxSession mfx_session;
    VADisplay va_dpy;
    VASurfaceID *surfaces;
    mfxMemId    *surface_ids;
    int         *surface_used;
    int       nb_surfaces;
    mfxFrameInfo frame_info;
 } DecodeContext;
 static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
                             mfxFrameAllocResponse *resp)
 {
    DecodeContext *decode = pthis;
    int err, i;
    if (decode->surfaces) {
        fprintf(stderr, "Multiple allocation requests.\n");
        return MFX_ERR_MEMORY_ALLOC;
    }
    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
        return MFX_ERR_UNSUPPORTED;
    }
    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
        fprintf(stderr, "Unsupported surface properties.\n");
        return MFX_ERR_UNSUPPORTED;
    }
    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
        goto fail;
    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
                           req->Info.Width, req->Info.Height,
                           decode->surfaces, req->NumFrameSuggested,
                           NULL, 0);
    if (err != VA_STATUS_SUCCESS) {
        fprintf(stderr, "Error allocating VA surfaces\n");
        goto fail;
    }
    decode->nb_surfaces = req->NumFrameSuggested;
    for (i = 0; i < decode->nb_surfaces; i++)
        decode->surface_ids[i] = &decode->surfaces[i];
    resp->mids           = decode->surface_ids;
    resp->NumFrameActual = decode->nb_surfaces;
    decode->frame_info = req->Info;
    return MFX_ERR_NONE;
 fail:
    av_freep(&decode->surfaces);
    av_freep(&decode->surface_ids);
    av_freep(&decode->surface_used);
    return MFX_ERR_MEMORY_ALLOC;
 }
 static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
 {
    DecodeContext *decode = pthis;
    if (decode->surfaces)
        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
    av_freep(&decode->surfaces);
    av_freep(&decode->surface_ids);
    av_freep(&decode->surface_used);
    decode->nb_surfaces = 0;
    return MFX_ERR_NONE;
 }
 static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
 {
    return MFX_ERR_UNSUPPORTED;
 }
 static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
 {
    return MFX_ERR_UNSUPPORTED;
 }
 static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
 {
    *hdl = mid;
    return MFX_ERR_NONE;
 }
 static void free_buffer(void *opaque, uint8_t *data)
 {
    int *used = opaque;
    *used = 0;
    av_freep(&data);
 }
 static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
 {
    DecodeContext *decode = avctx->opaque;
    mfxFrameSurface1 *surf;
    AVBufferRef *surf_buf;
    int idx;
    for (idx = 0; idx < decode->nb_surfaces; idx++) {
        if (!decode->surface_used[idx])
            break;
    }
    if (idx == decode->nb_surfaces) {
        fprintf(stderr, "No free surfaces\n");
        return AVERROR(ENOMEM);
    }
    surf = av_mallocz(sizeof(*surf));
    if (!surf)
        return AVERROR(ENOMEM);
    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
    if (!surf_buf) {
        av_freep(&surf);
        return AVERROR(ENOMEM);
    }
    surf->Info       = decode->frame_info;
    surf->Data.MemId = &decode->surfaces[idx];
    frame->buf[0]  = surf_buf;
    frame->data[3] = (uint8_t*)surf;
    decode->surface_used[idx] = 1;
    return 0;
 }
 static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
 {
    while (*pix_fmts != AV_PIX_FMT_NONE) {
        if (*pix_fmts == AV_PIX_FMT_QSV) {
            if (!avctx->hwaccel_context) {
                DecodeContext *decode = avctx->opaque;
                AVQSVContext *qsv = av_qsv_alloc_context();
                if (!qsv)
                    return AV_PIX_FMT_NONE;
                qsv->session   = decode->mfx_session;
                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
                avctx->hwaccel_context = qsv;
            }
            return AV_PIX_FMT_QSV;
        }
        pix_fmts++;
    }
    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
    return AV_PIX_FMT_NONE;
 }
 static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
                         AVFrame *frame, AVPacket *pkt,
                         AVIOContext *output_ctx)
 {
    int ret = 0;
    int got_frame = 1;
    while (pkt->size > 0 || (!pkt->data && got_frame)) {
        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
        if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            return ret;
        }
        pkt->data += ret;
        pkt->size -= ret;
        /* A real program would do something useful with the decoded frame here.
         * We just retrieve the raw data and write it to a file, which is rather
         * useless but pedagogic. */
        if (got_frame) {
            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
            VAImageFormat img_fmt = {
                .fourcc         = VA_FOURCC_NV12,
                .byte_order     = VA_LSB_FIRST,
                .bits_per_pixel = 8,
                .depth          = 8,
            };
            VAImage img;
            VAStatus err;
            uint8_t *data;
            int i, j;
            img.buf      = VA_INVALID_ID;
            img.image_id = VA_INVALID_ID;
            err = vaCreateImage(decode->va_dpy, &img_fmt,
                                frame->width, frame->height, &img);
            if (err != VA_STATUS_SUCCESS) {
                fprintf(stderr, "Error creating an image: %s\n",
                        vaErrorStr(err));
                ret = AVERROR_UNKNOWN;
                goto fail;
            }
            err = vaGetImage(decode->va_dpy, surface, 0, 0,
                             frame->width, frame->height,
                             img.image_id);
            if (err != VA_STATUS_SUCCESS) {
                fprintf(stderr, "Error getting an image: %s\n",
                        vaErrorStr(err));
                ret = AVERROR_UNKNOWN;
                goto fail;
            }
            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
            if (err != VA_STATUS_SUCCESS) {
                fprintf(stderr, "Error mapping the image buffer: %s\n",
                        vaErrorStr(err));
                ret = AVERROR_UNKNOWN;
                goto fail;
            }
            for (i = 0; i < img.num_planes; i++)
                for (j = 0; j < (img.height >> (i > 0)); j++)
                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
 fail:
            if (img.buf != VA_INVALID_ID)
                vaUnmapBuffer(decode->va_dpy, img.buf);
            if (img.image_id != VA_INVALID_ID)
                vaDestroyImage(decode->va_dpy, img.image_id);
            av_frame_unref(frame);
            if (ret < 0)
                return ret;
        }
    }
    return 0;
 }
 int main(int argc, char **argv)
 {
    AVFormatContext *input_ctx = NULL;
    AVStream *video_st = NULL;
    AVCodecContext *decoder_ctx = NULL;
    const AVCodec *decoder;
    AVPacket pkt = { 0 };
    AVFrame *frame = NULL;
    DecodeContext decode = { NULL };
    Display *dpy = NULL;
    int va_ver_major, va_ver_minor;
    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
    mfxVersion mfx_ver = { { 1, 1 } };
    mfxFrameAllocator frame_allocator = {
        .pthis = &decode,
        .Alloc = frame_alloc,
        .Lock  = frame_lock,
        .Unlock = frame_unlock,
        .GetHDL = frame_get_hdl,
        .Free   = frame_free,
    };
    AVIOContext *output_ctx = NULL;
    int ret, i, err;
    av_register_all();
    if (argc < 3) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        return 1;
    }
    /* open the input file */
    ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
    if (ret < 0) {
        fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
        goto finish;
    }
    /* find the first H.264 video stream */
    for (i = 0; i < input_ctx->nb_streams; i++) {
        AVStream *st = input_ctx->streams[i];
        if (st->codec->codec_id == AV_CODEC_ID_H264 && !video_st)
            video_st = st;
        else
            st->discard = AVDISCARD_ALL;
    }
    if (!video_st) {
        fprintf(stderr, "No H.264 video stream in the input file\n");
        goto finish;
    }
    /* initialize VA-API */
    dpy = XOpenDisplay(NULL);
    if (!dpy) {
        fprintf(stderr, "Cannot open the X display\n");
        goto finish;
    }
    decode.va_dpy = vaGetDisplay(dpy);
    if (!decode.va_dpy) {
        fprintf(stderr, "Cannot open the VA display\n");
        goto finish;
    }
    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
    if (err != VA_STATUS_SUCCESS) {
        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
        goto finish;
    }
    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
    /* initialize an MFX session */
    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
    if (err != MFX_ERR_NONE) {
        fprintf(stderr, "Error initializing an MFX session\n");
        goto finish;
    }
    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
    /* initialize the decoder */
    decoder = avcodec_find_decoder_by_name("h264_qsv");
    if (!decoder) {
        fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
        goto finish;
    }
    decoder_ctx = avcodec_alloc_context3(decoder);
    if (!decoder_ctx) {
        ret = AVERROR(ENOMEM);
        goto finish;
    }
    decoder_ctx->codec_id = AV_CODEC_ID_H264;
    if (video_st->codec->extradata_size) {
        decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
                                            FF_INPUT_BUFFER_PADDING_SIZE);
        if (!decoder_ctx->extradata) {
            ret = AVERROR(ENOMEM);
            goto finish;
        }
        memcpy(decoder_ctx->extradata, video_st->codec->extradata,
               video_st->codec->extradata_size);
        decoder_ctx->extradata_size = video_st->codec->extradata_size;
    }
    decoder_ctx->refcounted_frames = 1;
    decoder_ctx->opaque      = &decode;
    decoder_ctx->get_buffer2 = get_buffer;
    decoder_ctx->get_format  = get_format;
    ret = avcodec_open2(decoder_ctx, NULL, NULL);
    if (ret < 0) {
        fprintf(stderr, "Error opening the decoder: ");
        goto finish;
    }
    /* open the output stream */
    ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
    if (ret < 0) {
        fprintf(stderr, "Error opening the output context: ");
        goto finish;
    }
    frame = av_frame_alloc();
    if (!frame) {
        ret = AVERROR(ENOMEM);
        goto finish;
    }
    /* actual decoding */
    while (ret >= 0) {
        ret = av_read_frame(input_ctx, &pkt);
        if (ret < 0)
            break;
        if (pkt.stream_index == video_st->index)
            ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
        av_packet_unref(&pkt);
    }
    /* flush the decoder */
    pkt.data = NULL;
    pkt.size = 0;
    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
 finish:
    if (ret < 0) {
        char buf[1024];
        av_strerror(ret, buf, sizeof(buf));
        fprintf(stderr, "%s\n", buf);
    }
    avformat_close_input(&input_ctx);
    av_frame_free(&frame);
    if (decode.mfx_session)
        MFXClose(decode.mfx_session);
    if (decode.va_dpy)
        vaTerminate(decode.va_dpy);
    if (dpy)
        XCloseDisplay(dpy);
    if (decoder_ctx)
        av_freep(&decoder_ctx->hwaccel_context);
    avcodec_free_context(&decoder_ctx);
    avio_close(output_ctx);
    return ret;
 }
--- a/doc/examples/remuxing.c
+++ b/doc/examples/remuxing.c
@@ -1,165 +0,0 @@
 /*
 * 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;
        }
        out_stream->codec->codec_tag = 0;
        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_closep(&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;
 }
--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -1,214 +0,0 @@
 /*
 * Copyright (c) 2012 Stefano Sabatini
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @example resampling_audio.c
 * libswresample API use example.
 */
 #include <libavutil/opt.h>
 #include <libavutil/channel_layout.h>
 #include <libavutil/samplefmt.h>
 #include <libswresample/swresample.h>
 static int get_format_from_sample_fmt(const char **fmt,
                                      enum AVSampleFormat sample_fmt)
 {
    int i;
    struct sample_fmt_entry {
        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
    } sample_fmt_entries[] = {
        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
    };
    *fmt = NULL;
    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
        if (sample_fmt == entry->sample_fmt) {
            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
            return 0;
        }
    }
    fprintf(stderr,
            "Sample format %s not supported as output format\n",
            av_get_sample_fmt_name(sample_fmt));
    return AVERROR(EINVAL);
 }
 /**
 * Fill dst buffer with nb_samples, generated starting from t.
 */
 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;
    const double c = 2 * M_PI * 440.0;
    /* generate sin tone with 440Hz frequency and duplicated channels */
    for (i = 0; i < nb_samples; i++) {
        *dstp = sin(c * *t);
        for (j = 1; j < nb_channels; j++)
            dstp[j] = dstp[0];
        dstp += nb_channels;
        *t += tincr;
    }
 }
 int main(int argc, char **argv)
 {
    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
    int src_rate = 48000, dst_rate = 44100;
    uint8_t **src_data = NULL, **dst_data = NULL;
    int src_nb_channels = 0, dst_nb_channels = 0;
    int src_linesize, dst_linesize;
    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
    const char *dst_filename = NULL;
    FILE *dst_file;
    int dst_bufsize;
    const char *fmt;
    struct SwrContext *swr_ctx;
    double t;
    int ret;
    if (argc != 2) {
        fprintf(stderr, "Usage: %s output_file\n"
                "API example program to show how to resample an audio stream with libswresample.\n"
                "This program generates a series of audio frames, resamples them to a specified "
                "output format and rate and saves them to an output file named output_file.\n",
            argv[0]);
        exit(1);
    }
    dst_filename = argv[1];
    dst_file = fopen(dst_filename, "wb");
    if (!dst_file) {
        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
        exit(1);
    }
    /* create resampler context */
    swr_ctx = swr_alloc();
    if (!swr_ctx) {
        fprintf(stderr, "Could not allocate resampler context\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }
    /* set options */
    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
    /* initialize the resampling context */
    if ((ret = swr_init(swr_ctx)) < 0) {
        fprintf(stderr, "Failed to initialize the resampling context\n");
        goto end;
    }
    /* allocate source and destination samples buffers */
    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
                                             src_nb_samples, src_sample_fmt, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate source samples\n");
        goto end;
    }
    /* compute the number of converted samples: buffering is avoided
     * ensuring that the output buffer will contain at least all the
     * converted input samples */
    max_dst_nb_samples = dst_nb_samples =
        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
    /* buffer is going to be directly written to a rawaudio file, no alignment */
    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
                                             dst_nb_samples, dst_sample_fmt, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate destination samples\n");
        goto end;
    }
    t = 0;
    do {
        /* generate synthetic audio */
        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
        /* compute destination number of samples */
        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
        if (dst_nb_samples > max_dst_nb_samples) {
            av_freep(&dst_data[0]);
            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
                                   dst_nb_samples, dst_sample_fmt, 1);
            if (ret < 0)
                break;
            max_dst_nb_samples = dst_nb_samples;
        }
        /* convert to destination format */
        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
        if (ret < 0) {
            fprintf(stderr, "Error while converting\n");
            goto end;
        }
        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                 ret, dst_sample_fmt, 1);
        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);
    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
        goto end;
    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
 end:
    fclose(dst_file);
    if (src_data)
        av_freep(&src_data[0]);
    av_freep(&src_data);
    if (dst_data)
        av_freep(&dst_data[0]);
    av_freep(&dst_data);
    swr_free(&swr_ctx);
    return ret < 0;
 }
--- a/doc/examples/scaling_video.c
+++ b/doc/examples/scaling_video.c
@@ -1,140 +0,0 @@
 /*
 * Copyright (c) 2012 Stefano Sabatini
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * libswscale API use example.
 * @example scaling_video.c
 */
 #include <libavutil/imgutils.h>
 #include <libavutil/parseutils.h>
 #include <libswscale/swscale.h>
 static void fill_yuv_image(uint8_t *data[4], int linesize[4],
                           int width, int height, int frame_index)
 {
    int x, y;
    /* Y */
    for (y = 0; y < height; y++)
        for (x = 0; x < width; x++)
            data[0][y * linesize[0] + x] = x + y + frame_index * 3;
    /* Cb and Cr */
    for (y = 0; y < height / 2; y++) {
        for (x = 0; x < width / 2; x++) {
            data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
            data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
        }
    }
 }
 int main(int argc, char **argv)
 {
    uint8_t *src_data[4], *dst_data[4];
    int src_linesize[4], dst_linesize[4];
    int src_w = 320, src_h = 240, dst_w, dst_h;
    enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
    const char *dst_size = NULL;
    const char *dst_filename = NULL;
    FILE *dst_file;
    int dst_bufsize;
    struct SwsContext *sws_ctx;
    int i, ret;
    if (argc != 3) {
        fprintf(stderr, "Usage: %s output_file output_size\n"
                "API example program to show how to scale an image with libswscale.\n"
                "This program generates a series of pictures, rescales them to the given "
                "output_size and saves them to an output file named output_file\n."
                "\n", argv[0]);
        exit(1);
    }
    dst_filename = argv[1];
    dst_size     = argv[2];
    if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
        fprintf(stderr,
                "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
                dst_size);
        exit(1);
    }
    dst_file = fopen(dst_filename, "wb");
    if (!dst_file) {
        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
        exit(1);
    }
    /* create scaling context */
    sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
                             dst_w, dst_h, dst_pix_fmt,
                             SWS_BILINEAR, NULL, NULL, NULL);
    if (!sws_ctx) {
        fprintf(stderr,
                "Impossible to create scale context for the conversion "
                "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
                av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
                av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
        ret = AVERROR(EINVAL);
        goto end;
    }
    /* allocate source and destination image buffers */
    if ((ret = av_image_alloc(src_data, src_linesize,
                              src_w, src_h, src_pix_fmt, 16)) < 0) {
        fprintf(stderr, "Could not allocate source image\n");
        goto end;
    }
    /* buffer is going to be written to rawvideo file, no alignment */
    if ((ret = av_image_alloc(dst_data, dst_linesize,
                              dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
        fprintf(stderr, "Could not allocate destination image\n");
        goto end;
    }
    dst_bufsize = ret;
    for (i = 0; i < 100; i++) {
        /* generate synthetic video */
        fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
        /* convert to destination format */
        sws_scale(sws_ctx, (const uint8_t * const*)src_data,
                  src_linesize, 0, src_h, dst_data, dst_linesize);
        /* write scaled image to file */
        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
    }
    fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
           "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
           av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
 end:
    fclose(dst_file);
    av_freep(&src_data[0]);
    av_freep(&dst_data[0]);
    sws_freeContext(sws_ctx);
    return ret < 0;
 }
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -1,770 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 /**
 * @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 96000
 /** The number of output channels */
 #define OUTPUT_CHANNELS 2
 /**
 * Convert an error code into a text message.
 * @param error Error code to be converted
 * @return Corresponding error text (not thread-safe)
 */
 static const char *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 easier 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     = output_codec->sample_fmts[0];
    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
    /** Allow the use of the experimental AAC encoder */
    (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
    /** Set the sample rate for the container. */
    stream->time_base.den = input_codec_context->sample_rate;
    stream->time_base.num = 1;
    /**
     * 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_closep(&(*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, AVCodecContext *output_codec_context)
 {
    /** Create the FIFO buffer based on the specified output sample format. */
    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
                                      output_codec_context->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 at 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;
 }
 /** Global timestamp for the audio frames */
 static int64_t pts = 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);
    /** Set a timestamp based on the sample rate for the container. */
    if (frame) {
        frame->pts = pts;
        pts += frame->nb_samples;
    }
    /**
     * 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, output_codec_context))
        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_closep(&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;
 }
--- a/doc/examples/transcoding.c
+++ b/doc/examples/transcoding.c
@@ -1,583 +0,0 @@
 /*
 * Copyright (c) 2010 Nicolas George
 * Copyright (c) 2011 Stefano Sabatini
 * Copyright (c) 2014 Andrey Utkin
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * API example for demuxing, decoding, filtering, encoding and muxing
 * @example transcoding.c
 */
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
 #include <libavutil/pixdesc.h>
 static AVFormatContext *ifmt_ctx;
 static AVFormatContext *ofmt_ctx;
 typedef struct FilteringContext {
    AVFilterContext *buffersink_ctx;
    AVFilterContext *buffersrc_ctx;
    AVFilterGraph *filter_graph;
 } FilteringContext;
 static FilteringContext *filter_ctx;
 static int open_input_file(const char *filename)
 {
    int ret;
    unsigned int i;
    ifmt_ctx = NULL;
    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
        return ret;
    }
    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
        return ret;
    }
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        AVStream *stream;
        AVCodecContext *codec_ctx;
        stream = ifmt_ctx->streams[i];
        codec_ctx = stream->codec;
        /* Reencode video & audio and remux subtitles etc. */
        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
            /* Open decoder */
            ret = avcodec_open2(codec_ctx,
                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
                return ret;
            }
        }
    }
    av_dump_format(ifmt_ctx, 0, filename, 0);
    return 0;
 }
 static int open_output_file(const char *filename)
 {
    AVStream *out_stream;
    AVStream *in_stream;
    AVCodecContext *dec_ctx, *enc_ctx;
    AVCodec *encoder;
    int ret;
    unsigned int i;
    ofmt_ctx = NULL;
    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
    if (!ofmt_ctx) {
        av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
        return AVERROR_UNKNOWN;
    }
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        out_stream = avformat_new_stream(ofmt_ctx, NULL);
        if (!out_stream) {
            av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
            return AVERROR_UNKNOWN;
        }
        in_stream = ifmt_ctx->streams[i];
        dec_ctx = in_stream->codec;
        enc_ctx = out_stream->codec;
        if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
                || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
            /* in this example, we choose transcoding to same codec */
            encoder = avcodec_find_encoder(dec_ctx->codec_id);
            if (!encoder) {
                av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
                return AVERROR_INVALIDDATA;
            }
            /* In this example, we transcode to same properties (picture size,
             * sample rate etc.). These properties can be changed for output
             * streams easily using filters */
            if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
                enc_ctx->height = dec_ctx->height;
                enc_ctx->width = dec_ctx->width;
                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
                /* take first format from list of supported formats */
                enc_ctx->pix_fmt = encoder->pix_fmts[0];
                /* video time_base can be set to whatever is handy and supported by encoder */
                enc_ctx->time_base = dec_ctx->time_base;
            } else {
                enc_ctx->sample_rate = dec_ctx->sample_rate;
                enc_ctx->channel_layout = dec_ctx->channel_layout;
                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
                /* take first format from list of supported formats */
                enc_ctx->sample_fmt = encoder->sample_fmts[0];
                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
            }
            /* Third parameter can be used to pass settings to encoder */
            ret = avcodec_open2(enc_ctx, encoder, NULL);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
                return ret;
            }
        } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
            av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
            return AVERROR_INVALIDDATA;
        } else {
            /* if this stream must be remuxed */
            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
                    ifmt_ctx->streams[i]->codec);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
                return ret;
            }
        }
        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
    }
    av_dump_format(ofmt_ctx, 0, filename, 1);
    if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
        ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
            return ret;
        }
    }
    /* init muxer, write output file header */
    ret = avformat_write_header(ofmt_ctx, NULL);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
        return ret;
    }
    return 0;
 }
 static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
        AVCodecContext *enc_ctx, const char *filter_spec)
 {
    char args[512];
    int ret = 0;
    AVFilter *buffersrc = NULL;
    AVFilter *buffersink = NULL;
    AVFilterContext *buffersrc_ctx = NULL;
    AVFilterContext *buffersink_ctx = NULL;
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
    AVFilterGraph *filter_graph = avfilter_graph_alloc();
    if (!outputs || !inputs || !filter_graph) {
        ret = AVERROR(ENOMEM);
        goto end;
    }
    if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
        buffersrc = avfilter_get_by_name("buffer");
        buffersink = avfilter_get_by_name("buffersink");
        if (!buffersrc || !buffersink) {
            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
            ret = AVERROR_UNKNOWN;
            goto end;
        }
        snprintf(args, sizeof(args),
                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
                dec_ctx->time_base.num, dec_ctx->time_base.den,
                dec_ctx->sample_aspect_ratio.num,
                dec_ctx->sample_aspect_ratio.den);
        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                args, NULL, filter_graph);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
            goto end;
        }
        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
                NULL, NULL, filter_graph);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
            goto end;
        }
        ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
                (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
                AV_OPT_SEARCH_CHILDREN);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
            goto end;
        }
    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
        buffersrc = avfilter_get_by_name("abuffer");
        buffersink = avfilter_get_by_name("abuffersink");
        if (!buffersrc || !buffersink) {
            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
            ret = AVERROR_UNKNOWN;
            goto end;
        }
        if (!dec_ctx->channel_layout)
            dec_ctx->channel_layout =
                av_get_default_channel_layout(dec_ctx->channels);
        snprintf(args, sizeof(args),
                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
                av_get_sample_fmt_name(dec_ctx->sample_fmt),
                dec_ctx->channel_layout);
        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                args, NULL, filter_graph);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
            goto end;
        }
        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
                NULL, NULL, filter_graph);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
            goto end;
        }
        ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
                (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
                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_bin(buffersink_ctx, "channel_layouts",
                (uint8_t*)&enc_ctx->channel_layout,
                sizeof(enc_ctx->channel_layout), 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_bin(buffersink_ctx, "sample_rates",
                (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
                AV_OPT_SEARCH_CHILDREN);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
            goto end;
        }
    } else {
        ret = AVERROR_UNKNOWN;
        goto end;
    }
    /* Endpoints for the filter graph. */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
    outputs->pad_idx    = 0;
    outputs->next       = NULL;
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
    inputs->next       = NULL;
    if (!outputs->name || !inputs->name) {
        ret = AVERROR(ENOMEM);
        goto end;
    }
    if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
                    &inputs, &outputs, NULL)) < 0)
        goto end;
    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
        goto end;
    /* Fill FilteringContext */
    fctx->buffersrc_ctx = buffersrc_ctx;
    fctx->buffersink_ctx = buffersink_ctx;
    fctx->filter_graph = filter_graph;
 end:
    avfilter_inout_free(&inputs);
    avfilter_inout_free(&outputs);
    return ret;
 }
 static int init_filters(void)
 {
    const char *filter_spec;
    unsigned int i;
    int ret;
    filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
    if (!filter_ctx)
        return AVERROR(ENOMEM);
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        filter_ctx[i].buffersrc_ctx  = NULL;
        filter_ctx[i].buffersink_ctx = NULL;
        filter_ctx[i].filter_graph   = NULL;
        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
            continue;
        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
            filter_spec = "null"; /* passthrough (dummy) filter for video */
        else
            filter_spec = "anull"; /* passthrough (dummy) filter for audio */
        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
                ofmt_ctx->streams[i]->codec, filter_spec);
        if (ret)
            return ret;
    }
    return 0;
 }
 static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
    int ret;
    int got_frame_local;
    AVPacket enc_pkt;
    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
    if (!got_frame)
        got_frame = &got_frame_local;
    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
    /* encode filtered frame */
    enc_pkt.data = NULL;
    enc_pkt.size = 0;
    av_init_packet(&enc_pkt);
    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
            filt_frame, got_frame);
    av_frame_free(&filt_frame);
    if (ret < 0)
        return ret;
    if (!(*got_frame))
        return 0;
    /* prepare packet for muxing */
    enc_pkt.stream_index = stream_index;
    av_packet_rescale_ts(&enc_pkt,
                         ofmt_ctx->streams[stream_index]->codec->time_base,
                         ofmt_ctx->streams[stream_index]->time_base);
    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
    /* mux encoded frame */
    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
    return ret;
 }
 static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 {
    int ret;
    AVFrame *filt_frame;
    av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
    /* push the decoded frame into the filtergraph */
    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
            frame, 0);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
        return ret;
    }
    /* pull filtered frames from the filtergraph */
    while (1) {
        filt_frame = av_frame_alloc();
        if (!filt_frame) {
            ret = AVERROR(ENOMEM);
            break;
        }
        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
                filt_frame);
        if (ret < 0) {
            /* if no more frames for output - returns AVERROR(EAGAIN)
             * if flushed and no more frames for output - returns AVERROR_EOF
             * rewrite retcode to 0 to show it as normal procedure completion
             */
            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                ret = 0;
            av_frame_free(&filt_frame);
            break;
        }
        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
        ret = encode_write_frame(filt_frame, stream_index, NULL);
        if (ret < 0)
            break;
    }
    return ret;
 }
 static int flush_encoder(unsigned int stream_index)
 {
    int ret;
    int got_frame;
    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
                CODEC_CAP_DELAY))
        return 0;
    while (1) {
        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
        ret = encode_write_frame(NULL, stream_index, &got_frame);
        if (ret < 0)
            break;
        if (!got_frame)
            return 0;
    }
    return ret;
 }
 int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet = { .data = NULL, .size = 0 };
    AVFrame *frame = NULL;
    enum AVMediaType type;
    unsigned int stream_index;
    unsigned int i;
    int got_frame;
    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
    if (argc != 3) {
        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
        return 1;
    }
    av_register_all();
    avfilter_register_all();
    if ((ret = open_input_file(argv[1])) < 0)
        goto end;
    if ((ret = open_output_file(argv[2])) < 0)
        goto end;
    if ((ret = init_filters()) < 0)
        goto end;
    /* read all packets */
    while (1) {
        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
            break;
        stream_index = packet.stream_index;
        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
                stream_index);
        if (filter_ctx[stream_index].filter_graph) {
            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
            frame = av_frame_alloc();
            if (!frame) {
                ret = AVERROR(ENOMEM);
                break;
            }
            av_packet_rescale_ts(&packet,
                                 ifmt_ctx->streams[stream_index]->time_base,
                                 ifmt_ctx->streams[stream_index]->codec->time_base);
            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
                avcodec_decode_audio4;
            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
                    &got_frame, &packet);
            if (ret < 0) {
                av_frame_free(&frame);
                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
                break;
            }
            if (got_frame) {
                frame->pts = av_frame_get_best_effort_timestamp(frame);
                ret = filter_encode_write_frame(frame, stream_index);
                av_frame_free(&frame);
                if (ret < 0)
                    goto end;
            } else {
                av_frame_free(&frame);
            }
        } else {
            /* remux this frame without reencoding */
            av_packet_rescale_ts(&packet,
                                 ifmt_ctx->streams[stream_index]->time_base,
                                 ofmt_ctx->streams[stream_index]->time_base);
            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
            if (ret < 0)
                goto end;
        }
        av_free_packet(&packet);
    }
    /* flush filters and encoders */
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        /* flush filter */
        if (!filter_ctx[i].filter_graph)
            continue;
        ret = filter_encode_write_frame(NULL, i);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
            goto end;
        }
        /* flush encoder */
        ret = flush_encoder(i);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
            goto end;
        }
    }
    av_write_trailer(ofmt_ctx);
 end:
    av_free_packet(&packet);
    av_frame_free(&frame);
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        avcodec_close(ifmt_ctx->streams[i]->codec);
        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
            avcodec_close(ofmt_ctx->streams[i]->codec);
        if (filter_ctx && filter_ctx[i].filter_graph)
            avfilter_graph_free(&filter_ctx[i].filter_graph);
    }
    av_free(filter_ctx);
    avformat_close_input(&ifmt_ctx);
    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
        avio_closep(&ofmt_ctx->pb);
    avformat_free_context(ofmt_ctx);
    if (ret < 0)
        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
    return ret ? 1 : 0;
 }
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -1,17 +1,30 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg FAQ
@titlepage
@sp 7
@center @titlefont{FFmpeg FAQ}
@sp 3
@end titlepage
@top
@contents
@chapter General Questions
@section When will the next FFmpeg version be released? / Why are FFmpeg releases so few and far between?
 Like most open source projects FFmpeg suffers from a certain lack of
 manpower. For this reason the developers have to prioritize the work
 they do and putting out releases is not at the top of the list, fixing
 bugs and reviewing patches takes precedence. Please don't complain or
 request more timely and/or frequent releases unless you are willing to
 help out creating them.
@section I have a problem with an old version of FFmpeg; where should I report it?
 Nowhere. Upgrade to the latest release or if there is no recent release upgrade
 to Subversion HEAD. You could also try to report it. Maybe you will get lucky and
 become the first person in history to get an answer different from "upgrade
 to Subversion HEAD".
@section Why doesn't FFmpeg support feature [xyz]?
 Because no one has taken on that task yet. FFmpeg development is
@@ -25,6 +38,30 @@ No. Windows DLLs are not portable, bloated and often slow.
 Moreover FFmpeg strives to support all codecs natively.
 A DLL loader is not conducive to that goal.
@section My bug report/mail to ffmpeg-devel/user has not received any replies.
 Likely reasons
@itemize
@item We are busy and haven't had time yet to read your report or
 investigate the issue.
@item You didn't follow bugreports.html.
@item You didn't use Subversion HEAD.
@item You reported a segmentation fault without gdb output.
@item You describe a problem but not how to reproduce it.
@item It's unclear if you use ffmpeg as command line tool or use
 libav* from another application.
@item You speak about a video having problems on playback but
 not what you use to play it.
@item We have no faint clue what you are talking about besides
 that it is related to FFmpeg.
@end itemize
@section Is there a forum for FFmpeg? I do not like mailing lists.
 You may view our mailing lists with a more forum-alike look here:
@url{http://dir.gmane.org/gmane.comp.video.ffmpeg.user},
 but, if you post, please remember that our mailing list rules still apply there.
@section I cannot read this file although this format seems to be supported by ffmpeg.
 Even if ffmpeg can read the container format, it may not support all its
@@ -48,7 +85,7 @@ The following list of video codecs should work on most Windows systems:
 .asf only
@item mpeg4
 Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
-@item mpeg1video
+@item mpeg1
 .mpg only
@end table
 Note, ASF files often have .wmv or .wma extensions in Windows. It should also
@@ -60,9 +97,9 @@ The following list of audio codecs should work on most Windows systems:
@table @option
@item adpcm_ima_wav
@item adpcm_ms
-@item pcm_s16le
+@item pcm
 always
-@item libmp3lame
+@item mp3
 If some MP3 codec like LAME is installed.
@end table
@@ -80,73 +117,11 @@ not a bug they should fix:
 Then again, some of them do not know the difference between an undecidable
 problem and an NP-hard problem...
@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
 Distributions usually split libraries in several packages. The main package
 contains the files necessary to run programs using the library. The
 development package contains the files necessary to build programs using the
 library. Sometimes, docs and/or data are in a separate package too.
 To build FFmpeg, you need to install the development package. It is usually
 called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
 build is finished, but be sure to keep the main package.
@section How do I make @command{pkg-config} find my libraries?
 Somewhere along with your libraries, there is a @file{.pc} file (or several)
 in a @file{pkgconfig} directory. You need to set environment variables to
 point @command{pkg-config} to these files.
 If you need to @emph{add} directories to @command{pkg-config}'s search list
 (typical use case: library installed separately), add it to
@code{$PKG_CONFIG_PATH}:
@example
 export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
@end example
 If you need to @emph{replace} @command{pkg-config}'s search list
 (typical use case: cross-compiling), set it in
@code{$PKG_CONFIG_LIBDIR}:
@example
 export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
@end example
 If you need to know the library's internal dependencies (typical use: static
 linking), add the @code{--static} option to @command{pkg-config}:
@example
 ./configure --pkg-config-flags=--static
@end example
@section How do I use @command{pkg-config} when cross-compiling?
 The best way is to install @command{pkg-config} in your cross-compilation
 environment. It will automatically use the cross-compilation libraries.
 You can also use @command{pkg-config} from the host environment by
 specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
 In that case, you must point @command{pkg-config} to the correct directories
 using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
 As an intermediate solution, you can place in your cross-compilation
 environment a script that calls the host @command{pkg-config} with
@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
@example
 #!/bin/sh
 PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
 export PKG_CONFIG_LIBDIR
 exec /usr/bin/pkg-config "$@@"
@end example
@chapter Usage
@section ffmpeg does not work; what is wrong?
-Try a @code{make distclean} in the ffmpeg source directory before the build.
+Try a @code{make distclean} in the ffmpeg source directory before the build. If this does not help see
 If this does not help see
 (@url{http://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
@@ -156,21 +131,12 @@ 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.
-@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
+@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
 Use the @option{-start_number} option to declare a starting number for
 the sequence. This is useful if your sequence does not start with
@file{img001.jpg} but is still in a numerical order. The following
 example will start with @file{img100.jpg}:
@example
 ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example
 If you have large number of pictures to rename, you can use the
 following command to ease the burden. The command, using the bourne
@@ -179,7 +145,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
-x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+  x=1; for i in *jpg; do counter=$(printf %03d $x); ln "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
 If you want to sequence them by oldest modified first, substitute
@@ -188,23 +154,17 @@ 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.
 You can also use @command{cat} to pipe images to ffmpeg:
@example
 cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example
@section How do I encode movie to single pictures?
 Use:
@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
@@ -212,15 +172,15 @@ The @file{movie.mpg} used as input will be converted to
 Instead of relying on file format self-recognition, you may also use
@table @option
-@item -c:v ppm
+@item -vcodec ppm
-@item -c:v png
+@item -vcodec png
-@item -c:v mjpeg
+@item -vcodec mjpeg
@end table
 to force the encoding.
 Applying that to the previous example:
@example
-ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+  ffmpeg -i movie.mpg -f image2 -vcodec mjpeg menu%d.jpg
@end example
 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -235,28 +195,79 @@ quite logical that there is a small reduction of quality. This is not a bug.
 Use @file{-} as file name.
@section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux?
 This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at
 (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-420-2.4.26.patch?view=co}). This may also
 apply cleanly to other 2.4-series kernels.
@section How do I avoid the ugly aliasing artifacts in bt8x8 captures on Linux?
 Pass 'combfilter=1 lumafilter=1' to the bttv driver. Note though that 'combfilter=1'
 will cause somewhat too strong filtering. A fix is to apply (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.4.26.patch?view=co})
 or (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.6.6.patch?view=co})
 and pass 'combfilter=2'.
@section -f jpeg doesn't work.
 Try '-f image2 test%d.jpg'.
-@section Why can I not change the frame rate?
+@section Why can I not change the framerate?
-Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
+Some codecs, like MPEG-1/2, only allow a small number of fixed framerates.
-Choose a different codec with the -c:v command line option.
+Choose a different codec with the -vcodec command line option.
@section How do I encode Xvid or DivX video with ffmpeg?
 Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
 standard (note that there are many other coding formats that use this
-same standard). Thus, use '-c:v mpeg4' to encode in these formats. The
+same standard). Thus, use '-vcodec mpeg4' to encode in these formats. The
 default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
 a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
 force the fourcc 'xvid' to be stored as the video fourcc rather than the
 default.
@section How do I encode videos which play on the iPod?
@table @option
@item needed stuff
 -acodec libfaac -vcodec mpeg4 width<=320 height<=240
@item working stuff
 4mv, title
@item non-working stuff
 B-frames
@item example command line
 ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X output.mp4
@end table
@section How do I encode videos which play on the PSP?
@table @option
@item needed stuff
 -acodec libfaac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp
@item working stuff
 4mv, title
@item non-working stuff
 B-frames
@item example command line
 ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp output.mp4
@item needed stuff for H.264
 -acodec libfaac -vcodec libx264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
@item working stuff for H.264
 title, loop filter
@item non-working stuff for H.264
 CAVLC
@item example command line
 ffmpeg -i input -acodec libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
@item higher resolution for newer PSP firmwares, width<=480, height<=272
 -vcodec libx264 -level 21 -coder 1 -f psp
@item example command line
 ffmpeg -i input -acodec libfaac -ab 128kb -ac 2 -ar 48000 -vcodec libx264 -level 21 -b 640kb -coder 1 -f psp -flags +loop -trellis 2 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -s 480x272 output.mp4
@end table
@section Which are good parameters for encoding high quality MPEG-4?
-'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
+'-mbd rd -flags +4mv+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
 things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
@@ -275,82 +286,24 @@ material, and try '-top 0/1' if the result looks really messed-up.
 If you have built FFmpeg with @code{./configure --enable-avisynth}
 (only possible on MinGW/Cygwin platforms),
 then you may use any file that DirectShow can read as input.
 (Be aware that this feature has been recently added,
 so you will need to help yourself in case of problems.)
 Just create an "input.avs" text file with this single line ...
@example
-DirectShowSource("C:\path to your file\yourfile.asf")
+  DirectShowSource("C:\path to your file\yourfile.asf")
@end example
-... and then feed that text file to ffmpeg:
+... and then feed that text file to FFmpeg:
@example
-ffmpeg -i input.avs
+  ffmpeg -i input.avs
@end example
-For ANY other help on AviSynth, please visit the
+For ANY other help on Avisynth, please visit @url{http://www.avisynth.org/}.
@uref{http://www.avisynth.org/, AviSynth homepage}.
@section How can I join video files?
-To "join" video files is quite ambiguous. The following list explains the
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by
-different kinds of "joining" and points out how those are addressed in
+merely concatenating them.
 FFmpeg. To join video files may mean:
@itemize
@item
 To put them one after the other: this is called to @emph{concatenate} them
 (in short: concat) and is addressed
@ref{How can I concatenate video files, in this very faq}.
@item
 To put them together in the same file, to let the user choose between the
 different versions (example: different audio languages): this is called to
@emph{multiplex} them together (in short: mux), and is done by simply
 invoking ffmpeg with several @option{-i} options.
@item
 For audio, to put all channels together in a single stream (example: two
 mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
 For audio, to play one on top of the other: this is called to @emph{mix}
 them, and can be done by first merging them into a single stream and then
 using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
 the channels at will.
@item
 For video, to display both together, side by side or one on top of a part of
 the other; it can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@anchor{How can I concatenate video files}
@section How can I concatenate video files?
 There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
 FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
 documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
 FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
 your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
 FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
 documentation.
 A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
 video by merely concatenating the files containing them.
 Hence you may concatenate your multimedia files by first transcoding them to
 these privileged formats, then using the humble @code{cat} command (or the
@@ -358,38 +311,28 @@ equally humble @code{copy} under Windows), and finally transcoding back to your
 format of choice.
@example
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
+ffmpeg -i input1.avi -sameq intermediate1.mpg
-ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+ffmpeg -i input2.avi -sameq intermediate2.mpg
 cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+ffmpeg -i intermediate_all.mpg -sameq output.avi
@end example
-Additionally, you can use the @code{concat} protocol instead of @code{cat} or
+Notice that you should either use @code{-sameq} or set a reasonably high
-@code{copy} which will avoid creation of a potentially huge intermediate file.
+bitrate for your intermediate and output files, if you want to preserve
 video quality.
-@example
+Also notice that you may avoid the huge intermediate files by taking advantage
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
+of named pipes, should your platform support it:
 ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
 ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
 ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example
 Note that you may need to escape the character "|" which is special for many
 shells.
 Another option is usage of named pipes, should your platform support it:
@example
 mkfifo intermediate1.mpg
 mkfifo intermediate2.mpg
-ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
+ffmpeg -i input1.avi -sameq -y intermediate1.mpg < /dev/null &
-ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
+ffmpeg -i input2.avi -sameq -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
+ffmpeg -f mpeg -i - -sameq -vcodec mpeg4 -acodec libmp3lame output.avi
@end example
@subsection Concatenating using raw audio and video
 Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
 allow concatenation, and the transcoding step is almost lossless.
 When using multiple yuv4mpegpipe(s), the first line needs to be discarded
@@ -397,8 +340,7 @@ from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
 must use command grouping, @code{@{  ;@}}, to background properly.
-For example, let's say we want to concatenate two FLV files into an
+For example, let's say we want to join two FLV files into an output.flv file:
 output.flv file:
@example
 mkfifo temp1.a
@@ -415,104 +357,33 @@ cat temp1.a temp2.a > all.a &
 cat temp1.v temp2.v > all.v &
 ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
       -f yuv4mpegpipe -i all.v \
-       -y output.flv
+       -sameq -y output.flv
 rm temp[12].[av] all.[av]
@end example
-@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
+@section FFmpeg does not adhere to the -maxrate setting, some frames are bigger than maxrate/fps.
-Use @option{-dumpgraph -} to find out exactly where the channel layout is
+Read the MPEG spec about video buffer verifier.
 lost.
-Most likely, it is through @code{auto-inserted aresample}. Try to understand
+@section I want CBR, but no matter what I do frame sizes differ.
 why the converting filter was needed at that place.
-Just before the output is a likely place, as @option{-f lavfi} currently
+You do not understand what CBR is, please read the MPEG spec.
-only support packed S16.
+Read about video buffer verifier and constant bitrate.
 The one sentence summary is that there is a buffer and the input rate is
 constant, the output can vary as needed.
-Then insert the correct @code{aformat} explicitly in the filtergraph,
+@section How do I check if a stream is CBR?
 specifying the exact format.
-@example
+To quote the MPEG-2 spec:
-aformat=sample_fmts=s16:channel_layouts=stereo
+"There is no way to tell that a bitstream is constant bitrate without
-@end example
+examining all of the vbv_delay values and making complicated computations."
@section Why does FFmpeg not see the subtitles in my VOB file?
 VOB and a few other formats do not have a global header that describes
 everything present in the file. Instead, applications are supposed to scan
 the file to see what it contains. Since VOB files are frequently large, only
 the beginning is scanned. If the subtitles happen only later in the file,
 they will not be initially detected.
 Some applications, including the @code{ffmpeg} command-line tool, can only
 work with streams that were detected during the initial scan; streams that
 are detected later are ignored.
 The size of the initial scan is controlled by two options: @code{probesize}
 (default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
 the subtitle stream to be detected, both values must be large enough.
@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
 The @option{-sameq} option meant "same quantizer", and made sense only in a
 very limited set of cases. Unfortunately, a lot of people mistook it for
 "same quality" and used it in places where it did not make sense: it had
 roughly the expected visible effect, but achieved it in a very inefficient
 way.
 Each encoder has its own set of options to set the quality-vs-size balance,
 use the options for the encoder you are using to set the quality level to a
 point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
 of the encoder you chose.
@section I have a stretched video, why does scaling does not fix it?
 A lot of video codecs and formats can store the @emph{aspect ratio} of the
 video: this is the ratio between the width and the height of either the full
 image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
 ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
 SAR.
 Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
 of video standards, especially from the analogic-numeric transition era, use
 non-square pixels.
 Most processing filters in FFmpeg handle the aspect ratio to avoid
 stretching the image: cropping adjusts the DAR to keep the SAR constant,
 scaling adjusts the SAR to keep the DAR constant.
 If you want to stretch, or “unstretch”, the image, you need to override the
 information with the
@url{http://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
 Do not forget to examine carefully the original video to check whether the
 stretching comes from the image or from the aspect ratio information.
 For example, to fix a badly encoded EGA capture, use the following commands,
 either the first one to upscale to square pixels or the second one to set
 the correct aspect ratio or the third one to avoid transcoding (may not work
 depending on the format / codec / player / phase of the moon):
@example
 ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
 ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
 ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
@end example
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
-Yes. Check the @file{doc/examples} directory in the source
+Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively,
 repository, also available online at:
@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
 Examples are also installed by default, usually in
@code{$PREFIX/share/ffmpeg/examples}.
 Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
 examine the source code for one of the many open source projects that
 already incorporate FFmpeg at (@url{projects.html}).
@@ -524,52 +395,64 @@ with @code{#ifdef}s related to the compiler.
@section Is Microsoft Visual C++ supported?
-Yes. Please see the @uref{platform.html, Microsoft Visual C++}
+No. Microsoft Visual C++ is not compliant to the C99 standard and does
-section in the FFmpeg documentation.
+not - among other things - support the inline assembly used in FFmpeg.
 If you wish to use MSVC++ for your
 project then you can link the MSVC++ code with libav* as long as
 you compile the latter with a working C compiler. For more information, see
 the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
 documentation.
 There have been efforts to make FFmpeg compatible with MSVC++ in the
 past. However, they have all been rejected as too intrusive, especially
 since MinGW does the job adequately. None of the core developers
 work with MSVC++ and thus this item is low priority. Should you find
 the silver bullet that solves this problem, feel free to shoot it at us.
 We strongly recommend you to move over from MSVC++ to MinGW tools.
@section Can I use FFmpeg or libavcodec under Windows?
 Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
 Read the @emph{Windows} section in the FFmpeg documentation to find more
 information.
 To get help and instructions for building FFmpeg under Windows, check out
 the FFmpeg Windows Help Forum at
@url{http://ffmpeg.arrozcru.org/}.
@section Can you add automake, libtool or autoconf support?
 No. These tools are too bloated and they complicate the build.
-@section Why not rewrite FFmpeg in object-oriented C++?
+@section Why not rewrite ffmpeg in object-oriented C++?
 FFmpeg is already organized in a highly modular manner and does not need to
 be rewritten in a formal object language. Further, many of the developers
 favor straight C; it works for them. For more arguments on this matter,
-read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
+read "Programming Religion" at (@url{http://www.tux.org/lkml/#s15}).
@section Why are the ffmpeg programs devoid of debugging symbols?
-The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
+The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
-contain full debug information. Those binaries are stripped to create
+information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
-@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
+you need the debug information, used the *_g versions.
 the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?
 Yes, as long as the code is optional and can easily and cleanly be placed
-under #if CONFIG_GPL without breaking anything. So, for example, a new codec
+under #if CONFIG_GPL without breaking anything. So for example a new codec
 or filter would be OK under GPL while a bug fix to LGPL code would not.
-@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.
+@section I want to compile xyz.c alone but my compiler produced many errors.
-FFmpeg builds static libraries by default. In static libraries, dependencies
+Common code is in its own files in libav* and is used by the individual
-are not handled. That has two consequences. First, you must specify the
+codecs. They will not work without the common parts, you have to compile
-libraries in dependency order: @code{-lavdevice} must come before
+the whole libav*. If you wish, disable some parts with configure switches.
-@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
+You can also try to hack it and remove more, but if you had problems fixing
-Second, external libraries that are used in FFmpeg have to be specified too.
+the compilation failure then you are probably not qualified for this.
-An easy way to get the full list of required libraries in dependency order
+@section I'm using libavcodec from within my C++ application but the linker complains about missing symbols which seem to be available.
 is to use @code{pkg-config}.
@example
 c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example
 See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
 more details.
@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.
 FFmpeg is a pure C project, so to use the libraries within your C++ application
 you need to explicitly state that you are using a C library. You can do this by
@@ -577,19 +460,31 @@ encompassing your FFmpeg includes using @code{extern "C"}.
 See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
 FFmpeg is a pure C project using C99 math features, in order to enable C++
 to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
-You have to create a custom AVIOContext using @code{avio_alloc_context},
+You have to implement a URLProtocol, see libavformat/file.c in FFmpeg
-see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
+and libmpdemux/demux_lavf.c in MPlayer sources.
@section I get "No compatible shell script interpreter found." in MSys.
 The standard MSys bash (2.04) is broken. You need to install 2.05 or later.
@section I get "./configure: line <xxx>: pr: command not found" in MSys.
 The standard MSys install doesn't come with pr. You need to get it from the coreutils package.
@section I tried to pass RTP packets into a decoder, but it doesn't work.
 RTP is a container format like any other, you must first depacketize the
 codec frames/samples stored in RTP and then feed to the decoder.
@section Where can I find libav* headers for Pascal/Delphi?
 see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
-see @url{http://www.ffmpeg.org/~michael/}
+see @url{http://svn.ffmpeg.org/michael/trunk/docs/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
@@ -597,25 +492,12 @@ Even if peculiar since it is network oriented, RTP is a container like any
 other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
 In this specific case please look at RFC 4629 to see how it should be done.
-@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
+@section AVStream.r_frame_rate is wrong, it is much larger than the framerate.
-@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
+r_frame_rate is NOT the average framerate, it is the smallest framerate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
+For example, if you have mixed 25 and 30 fps content, then r_frame_rate
-will be 150 (it is the least common multiple).
+will be 150.
 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?
 Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
 or @code{FATE_SAMPLES} environment variable or the @code{--samples}
@command{configure} option is set to the right path.
@section Why is @code{make fate} not finding the samples?
 Do you happen to have a @code{~} character in the samples path to indicate a
 home directory? The value is used in ways where the shell cannot expand it,
 causing FATE to not find files. Just replace @code{~} by the full path.
@bye
--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -1,206 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Automated Testing Environment
@titlepage
@center @titlefont{FFmpeg Automated Testing Environment}
@end titlepage
@node Top
@top
@contents
@chapter Introduction
 FATE is an extended regression suite on the client-side and a means
 for results aggregation and presentation on the server-side.
 The first part of this document explains how you can use FATE from
 your FFmpeg source directory to test your ffmpeg binary. The second
 part describes how you can run FATE to submit the results to FFmpeg's
 FATE server.
 In any way you can have a look at the publicly viewable FATE results
 by visiting this website:
@url{http://fate.ffmpeg.org/}
 This is especially recommended for all people contributing source
 code to FFmpeg, as it can be seen if some test on some platform broke
 with 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
 submit your results to FFmpeg's FATE server. If you want to submit your
 results be sure to check that your combination of CPU, OS and compiler
 is not already listed on the above mentioned website.
 In the third part you can find a comprehensive listing of FATE makefile
 targets and variables.
@chapter Using FATE from your FFmpeg source directory
 If you want to run FATE on your machine you need to have the samples
 in place. You can get the samples via the build target fate-rsync.
 Use this command from the top-level source directory:
@example
 make fate-rsync SAMPLES=fate-suite/
 make fate       SAMPLES=fate-suite/
@end example
 The above commands set the samples location by passing a makefile
 variable via command line. It is also possible to set the samples
 location at source configuration time by invoking configure with
@option{--samples=<path to the samples directory>}. Afterwards you can
 invoke the makefile targets without setting the @var{SAMPLES} makefile
 variable. This is illustrated by the following commands:
@example
 ./configure --samples=fate-suite/
 make fate-rsync
 make fate
@end example
 Yet another way to tell FATE about the location of the sample
 directory is by making sure the environment variable FATE_SAMPLES
 contains the path to your samples directory. This can be achieved
 by e.g. putting that variable in your shell profile or by setting
 it in your interactive session.
@example
 FATE_SAMPLES=fate-suite/ make fate
@end example
@float NOTE
 Do not put a '~' character in the samples path to indicate a home
 directory. Because of shell nuances, this will cause FATE to fail.
@end float
 To use a custom wrapper to run the test, pass @option{--target-exec} to
@command{configure} or set the @var{TARGET_EXEC} Make variable.
@chapter Submitting the results to the FFmpeg result aggregation server
 To submit your results to the server you should run fate through the
 shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.
@example
 tests/fate.sh /path/to/fate_config
@end example
 A configuration file template with comments describing the individual
 configuration variables can be found at @file{doc/fate_config.sh.template}.
@ifhtml
 The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
 Create a configuration that suits your needs, based on the configuration
 template. The @env{slot} configuration variable can be any string that is not
 yet used, but it is suggested that you name it adhering to the following
 pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
 configuration file itself will be sourced in a shell script, therefore all
 shell features may be used. This enables you to setup the environment as you
 need it for your build.
 For your first test runs the @env{fate_recv} variable should be empty or
 commented out. This will run everything as normal except that it will omit
 the submission of the results to the server. The following files should be
 present in $workdir as specified in the configuration file:
@itemize
    @item configure.log
    @item compile.log
    @item test.log
    @item report
    @item version
@end itemize
 When you have everything working properly you can create an SSH key pair
 and send the public key to the FATE server administrator who can be contacted
 at the email address @email{fate-admin@@ffmpeg.org}.
 Configure your SSH client to use public key authentication with that key
 when connecting to the FATE server. Also do not forget to check the identity
 of the server and to accept its host key. This can usually be achieved by
 running your SSH client manually and killing it after you accepted the key.
 The FATE server's fingerprint is:
@table @samp
@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
 get detailed output concerning your SSH configuration and the authentication
 process.
 The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.
@chapter FATE makefile targets and variables
@section Makefile targets
@table @option
@item fate-rsync
 Download/synchronize sample files to the configured samples directory.
@item fate-list
 Will list all fate/regression test targets.
@item fate
 Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@table @env
@item V
 Verbosity level, can be set to 0, 1 or 2.
    @itemize
        @item 0: show just the test arguments
        @item 1: show just the command used in the test
        @item 2: show everything
    @end itemize
@item SAMPLES
 Specify or override the path to the FATE samples at make time, it has a
 meaning only while running the regression tests.
@item THREADS
 Specify how many threads to use while running regression tests, it is
 quite useful to detect thread-related regressions.
@item THREAD_TYPE
 Specify which threading strategy test, either @samp{slice} or @samp{frame},
 by default @samp{slice+frame}
@item CPUFLAGS
 Specify CPU flags.
@item TARGET_EXEC
 Specify or override the wrapper used to run the tests.
 The @env{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 @samp{1} to generate the missing or mismatched references.
@end table
@section Examples
@example
 make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
@end example
--- a/doc/fate_config.sh.template
+++ b/doc/fate_config.sh.template
@@ -1,30 +0,0 @@
 slot=                                    # some unique identifier
 repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
 #branch=release/2.6                       # the branch to test
 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=
 extra_conf=     # extra configure options not covered above
 #make=          # name of GNU make if not 'make'
 makeopts=       # extra options passed to 'make'
 #tar=           # command to create a tar archive from its arguments on stdout,
                # defaults to 'tar c'
--- a/doc/ffmpeg-bitstream-filters.texi
+++ b/doc/ffmpeg-bitstream-filters.texi
@@ -1,46 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Bitstream Filters Documentation
@titlepage
@center @titlefont{FFmpeg Bitstream Filters Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 This document describes the bitstream filters provided by the
 libavcodec library.
 A bitstream filter operates on the encoded stream data, and performs
 bitstream level modifications without performing decoding.
@c man end DESCRIPTION
@include bitstream_filters.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-bitstream-filters
@settitle FFmpeg bitstream filters
@end ignore
@bye
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
@@ -1,43 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Codecs Documentation
@titlepage
@center @titlefont{FFmpeg Codecs Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 This document describes the codecs (decoders and encoders) provided by
 the libavcodec library.
@c man end DESCRIPTION
@include codecs.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-codecs
@settitle FFmpeg codecs
@end ignore
@bye
--- a/doc/ffmpeg-devices.texi
+++ b/doc/ffmpeg-devices.texi
@@ -1,43 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Devices Documentation
@titlepage
@center @titlefont{FFmpeg Devices Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 This document describes the input and output devices provided by the
 libavdevice library.
@c man end DESCRIPTION
@include devices.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-devices
@settitle FFmpeg devices
@end ignore
@bye
--- a/doc/ffmpeg-doc.texi
+++ b/doc/ffmpeg-doc.texi
@@ -0,0 +1,956 @@
 \input texinfo @c -*- texinfo -*-
@settitle FFmpeg Documentation
@titlepage
@sp 7
@center @titlefont{FFmpeg Documentation}
@sp 3
@end titlepage
@chapter Introduction
 FFmpeg is a very fast video and audio converter. It can also grab from
 a live audio/video source.
 The command line interface is designed to be intuitive, in the sense
 that FFmpeg tries to figure out all parameters that can possibly be
 derived automatically. You usually only have to specify the target
 bitrate you want.
 FFmpeg can also convert from any sample rate to any other, and resize
 video on the fly with a high quality polyphase filter.
@chapter Quick Start
@c man begin EXAMPLES
@section Video and Audio grabbing
 FFmpeg can grab video and audio from devices given that you specify the input
 format and device.
@example
 ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
@end example
 Note that you must activate the right video source and channel before
 launching FFmpeg with any TV viewer such as xawtv
 (@url{http://linux.bytesex.org/xawtv/}) by Gerd Knorr. You also
 have to set the audio recording levels correctly with a
 standard mixer.
@section X11 grabbing
 FFmpeg can grab the X11 display.
@example
 ffmpeg -f x11grab -s cif -i :0.0 /tmp/out.mpg
@end example
 0.0 is display.screen number of your X11 server, same as
 the DISPLAY environment variable.
@example
 ffmpeg -f x11grab -s cif -i :0.0+10,20 /tmp/out.mpg
@end example
 0.0 is display.screen number of your X11 server, same as the DISPLAY environment
 variable. 10 is the x-offset and 20 the y-offset for the grabbing.
@section Video and Audio file format conversion
 * FFmpeg can use any supported file format and protocol as input:
 Examples:
 * You can use YUV files as input:
@example
 ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
@end example
 It will use the files:
@example
 /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
 /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
@end example
 The Y files use twice the resolution of the U and V files. They are
 raw files, without header. They can be generated by all decent video
 decoders. You must specify the size of the image with the @option{-s} option
 if FFmpeg cannot guess it.
 * You can input from a raw YUV420P file:
@example
 ffmpeg -i /tmp/test.yuv /tmp/out.avi
@end example
 test.yuv is a file containing raw YUV planar data. Each frame is composed
 of the Y plane followed by the U and V planes at half vertical and
 horizontal resolution.
 * You can output to a raw YUV420P file:
@example
 ffmpeg -i mydivx.avi hugefile.yuv
@end example
 * You can set several input files and output files:
@example
 ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
@end example
 Converts the audio file a.wav and the raw YUV video file a.yuv
 to MPEG file a.mpg.
 * You can also do audio and video conversions at the same time:
@example
 ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
@end example
 Converts a.wav to MPEG audio at 22050 Hz sample rate.
 * You can encode to several formats at the same time and define a
 mapping from input stream to output streams:
@example
 ffmpeg -i /tmp/a.wav -ab 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
@end example
 Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map
 file:index' specifies which input stream is used for each output
 stream, in the order of the definition of output streams.
 * You can transcode decrypted VOBs:
@example
 ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
@end example
 This is a typical DVD ripping example; the input is a VOB file, the
 output an AVI file with MPEG-4 video and MP3 audio. Note that in this
 command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
 GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
 input video. Furthermore, the audio stream is MP3-encoded so you need
 to enable LAME support by passing @code{--enable-libmp3lame} to configure.
 The mapping is particularly useful for DVD transcoding
 to get the desired audio language.
 NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
 * You can extract images from a video, or create a video from many images:
 For extracting images from a video:
@example
 ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
@end example
 This will extract one video frame per second from the video and will
 output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
 etc. Images will be rescaled to fit the new WxH values.
 If you want to extract just a limited number of frames, you can use the
 above command in combination with the -vframes or -t option, or in
 combination with -ss to start extracting from a certain point in time.
 For creating a video from many images:
@example
 ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example
 The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
 composed of three digits padded with zeroes to express the sequence
 number. It is the same syntax supported by the C printf function, but
 only formats accepting a normal integer are suitable.
 * You can put many streams of the same type in the output:
@example
 ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
@end example
 In addition to the first video and audio streams, the resulting
 output file @file{test12.avi} will contain the second video
 and the second audio stream found in the input streams list.
 The @code{-newvideo}, @code{-newaudio} and @code{-newsubtitle}
 options have to be specified immediately after the name of the output
 file to which you want to add them.
@c man end
@chapter Invocation
@section Syntax
 The generic syntax is:
@example
@c man begin SYNOPSIS
 ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
@c man end
@end example
@c man begin DESCRIPTION
 As a general rule, options are applied to the next specified
 file. Therefore, order is important, and you can have the same
 option on the command line multiple times. Each occurrence is
 then applied to the next input or output file.
 * To set the video bitrate of the output file to 64kbit/s:
@example
 ffmpeg -i input.avi -b 64k output.avi
@end example
 * To force the frame rate of the output file to 24 fps:
@example
 ffmpeg -i input.avi -r 24 output.avi
@end example
 * To force the frame rate of the input file (valid for raw formats only)
 to 1 fps and the frame rate of the output file to 24 fps:
@example
 ffmpeg -r 1 -i input.m2v -r 24 output.avi
@end example
 The format option may be needed for raw input files.
 By default, FFmpeg tries to convert as losslessly as possible: It
 uses the same audio and video parameters for the outputs as the one
 specified for the inputs.
@c man end
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -f @var{fmt}
 Force format.
@item -i @var{filename}
 input file name
@item -y
 Overwrite output files.
@item -t @var{duration}
 Restrict the transcoded/captured video sequence
 to the duration specified in seconds.
@code{hh:mm:ss[.xxx]} syntax is also supported.
@item -fs @var{limit_size}
 Set the file size limit.
@item -ss @var{position}
 Seek to given time position in seconds.
@code{hh:mm:ss[.xxx]} syntax is also supported.
@item -itsoffset @var{offset}
 Set the input time offset in seconds.
@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
 This option affects all the input files that follow it.
 The offset is added to the timestamps of the input files.
 Specifying a positive offset means that the corresponding
 streams are delayed by 'offset' seconds.
@item -timestamp @var{time}
 Set the timestamp.
@item -metadata @var{key}=@var{value}
 Set a metadata key/value pair.
 For example, for setting the title in the output file:
@example
 ffmpeg -i in.avi -metadata title="my title" out.flv
@end example
@item -v @var{number}
 Set the logging verbosity level.
@item -target @var{type}
 Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd",
 "ntsc-svcd", ... ). All the format options (bitrate, codecs,
 buffer sizes) are then set automatically. You can just type:
@example
 ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
@end example
 Nevertheless you can specify additional options as long as you know
 they do not conflict with the standard, as in:
@example
 ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dframes @var{number}
 Set the number of data frames to record.
@item -scodec @var{codec}
 Force subtitle codec ('copy' to copy stream).
@item -newsubtitle
 Add a new subtitle stream to the current output stream.
@item -slang @var{code}
 Set the ISO 639 language code (3 letters) of the current subtitle stream.
@end table
@section Video Options
@table @option
@item -b @var{bitrate}
 Set the video bitrate in bit/s (default = 200 kb/s).
@item -vframes @var{number}
 Set the number of video frames to record.
@item -r @var{fps}
 Set frame rate (Hz value, fraction or abbreviation), (default = 25).
@item -s @var{size}
 Set frame size. The format is @samp{wxh} (ffserver default = 160x128, ffmpeg default = same as source).
 The following abbreviations are recognized:
@table @samp
@item sqcif
 128x96
@item qcif
 176x144
@item cif
 352x288
@item 4cif
 704x576
@item 16cif
 1408x1152
@item qqvga
 160x120
@item qvga
 320x240
@item vga
 640x480
@item svga
 800x600
@item xga
 1024x768
@item uxga
 1600x1200
@item qxga
 2048x1536
@item sxga
 1280x1024
@item qsxga
 2560x2048
@item hsxga
 5120x4096
@item wvga
 852x480
@item wxga
 1366x768
@item wsxga
 1600x1024
@item wuxga
 1920x1200
@item woxga
 2560x1600
@item wqsxga
 3200x2048
@item wquxga
 3840x2400
@item whsxga
 6400x4096
@item whuxga
 7680x4800
@item cga
 320x200
@item ega
 640x350
@item hd480
 852x480
@item hd720
 1280x720
@item hd1080
 1920x1080
@end table
@item -aspect @var{aspect}
 Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777).
@item -croptop @var{size}
 Set top crop band size (in pixels).
@item -cropbottom @var{size}
 Set bottom crop band size (in pixels).
@item -cropleft @var{size}
 Set left crop band size (in pixels).
@item -cropright @var{size}
 Set right crop band size (in pixels).
@item -padtop @var{size}
 Set top pad band size (in pixels).
@item -padbottom @var{size}
 Set bottom pad band size (in pixels).
@item -padleft @var{size}
 Set left pad band size (in pixels).
@item -padright @var{size}
 Set right pad band size (in pixels).
@item -padcolor @var{hex_color}
 Set color of padded bands. The value for padcolor is expressed
 as a six digit hexadecimal number where the first two digits
 represent red, the middle two digits green and last two digits
 blue (default = 000000 (black)).
@item -vn
 Disable video recording.
@item -bt @var{tolerance}
 Set video bitrate tolerance (in bits, default 4000k).
 Has a minimum value of: (target_bitrate/target_framerate).
 In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
 willing to deviate from the target average bitrate value. This is
 not related to min/max bitrate. Lowering tolerance too much has
 an adverse effect on quality.
@item -maxrate @var{bitrate}
 Set max video bitrate (in bit/s).
 Requires -bufsize to be set.
@item -minrate @var{bitrate}
 Set min video bitrate (in bit/s).
 Most useful in setting up a CBR encode:
@example
 ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
@end example
 It is of little use elsewise.
@item -bufsize @var{size}
 Set video buffer verifier buffer size (in bits).
@item -vcodec @var{codec}
 Force video codec to @var{codec}. Use the @code{copy} special value to
 tell that the raw codec data must be copied as is.
@item -sameq
 Use same video quality as source (implies VBR).
@item -pass @var{n}
 Select the pass number (1 or 2). It is used to do two-pass
 video encoding. The statistics of the video are recorded in the first
 pass into a log file (see also the option -passlogfile),
 and in the second pass that log file is used to generate the video
 at the exact requested bitrate.
 On pass 1, you may just deactivate audio and set output to null,
 examples for Windows and Unix:
@example
 ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL
 ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
@end example
@item -passlogfile @var{prefix}
 Set two-pass log file name prefix to @var{prefix}, the default file name
 prefix is ``ffmpeg2pass''. The complete file name will be
@file{PREFIX-N.log}, where N is a number specific to the output
 stream.
@item -newvideo
 Add a new video stream to the current output stream.
@item -vlang @var{code}
 Set the ISO 639 language code (3 letters) of the current video stream.
@end table
@section Advanced Video Options
@table @option
@item -pix_fmt @var{format}
 Set pixel format. Use 'list' as parameter to show all the supported
 pixel formats.
@item -sws_flags @var{flags}
 Set SwScaler flags.
@item -g @var{gop_size}
 Set the group of pictures size.
@item -intra
 Use only intra frames.
@item -vdt @var{n}
 Discard threshold.
@item -qscale @var{q}
 Use fixed video quantizer scale (VBR).
@item -qmin @var{q}
 minimum video quantizer scale (VBR)
@item -qmax @var{q}
 maximum video quantizer scale (VBR)
@item -qdiff @var{q}
 maximum difference between the quantizer scales (VBR)
@item -qblur @var{blur}
 video quantizer scale blur (VBR) (range 0.0 - 1.0)
@item -qcomp @var{compression}
 video quantizer scale compression (VBR) (default 0.5).
 Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
@item -lmin @var{lambda}
 minimum video lagrange factor (VBR)
@item -lmax @var{lambda}
 max video lagrange factor (VBR)
@item -mblmin @var{lambda}
 minimum macroblock quantizer scale (VBR)
@item -mblmax @var{lambda}
 maximum macroblock quantizer scale (VBR)
 These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
 but you may use the QP2LAMBDA constant to easily convert from 'q' units:
@example
 ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end example
@item -rc_init_cplx @var{complexity}
 initial complexity for single pass encoding
@item -b_qfactor @var{factor}
 qp factor between P- and B-frames
@item -i_qfactor @var{factor}
 qp factor between P- and I-frames
@item -b_qoffset @var{offset}
 qp offset between P- and B-frames
@item -i_qoffset @var{offset}
 qp offset between P- and I-frames
@item -rc_eq @var{equation}
 Set rate control equation (@pxref{FFmpeg formula
 evaluator}) (default = @code{tex^qComp}).
@item -rc_override @var{override}
 rate control override for specific intervals
@item -me_method @var{method}
 Set motion estimation method to @var{method}.
 Available methods are (from lowest to best quality):
@table @samp
@item zero
 Try just the (0, 0) vector.
@item phods
@item log
@item x1
@item hex
@item umh
@item epzs
 (default method)
@item full
 exhaustive search (slow and marginally better than epzs)
@end table
@item -dct_algo @var{algo}
 Set DCT algorithm to @var{algo}. Available values are:
@table @samp
@item 0
 FF_DCT_AUTO (default)
@item 1
 FF_DCT_FASTINT
@item 2
 FF_DCT_INT
@item 3
 FF_DCT_MMX
@item 4
 FF_DCT_MLIB
@item 5
 FF_DCT_ALTIVEC
@end table
@item -idct_algo @var{algo}
 Set IDCT algorithm to @var{algo}. Available values are:
@table @samp
@item 0
 FF_IDCT_AUTO (default)
@item 1
 FF_IDCT_INT
@item 2
 FF_IDCT_SIMPLE
@item 3
 FF_IDCT_SIMPLEMMX
@item 4
 FF_IDCT_LIBMPEG2MMX
@item 5
 FF_IDCT_PS2
@item 6
 FF_IDCT_MLIB
@item 7
 FF_IDCT_ARM
@item 8
 FF_IDCT_ALTIVEC
@item 9
 FF_IDCT_SH4
@item 10
 FF_IDCT_SIMPLEARM
@end table
@item -er @var{n}
 Set error resilience to @var{n}.
@table @samp
@item 1
 FF_ER_CAREFUL (default)
@item 2
 FF_ER_COMPLIANT
@item 3
 FF_ER_AGGRESSIVE
@item 4
 FF_ER_VERY_AGGRESSIVE
@end table
@item -ec @var{bit_mask}
 Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
 the following values:
@table @samp
@item 1
 FF_EC_GUESS_MVS (default = enabled)
@item 2
 FF_EC_DEBLOCK (default = enabled)
@end table
@item -bf @var{frames}
 Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
@item -mbd @var{mode}
 macroblock decision
@table @samp
@item 0
 FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).
@item 1
 FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
@item 2
 FF_MB_DECISION_RD: rate distortion
@end table
@item -4mv
 Use four motion vector by macroblock (MPEG-4 only).
@item -part
 Use data partitioning (MPEG-4 only).
@item -bug @var{param}
 Work around encoder bugs that are not auto-detected.
@item -strict @var{strictness}
 How strictly to follow the standards.
@item -aic
 Enable Advanced intra coding (h263+).
@item -umv
 Enable Unlimited Motion Vector (h263+)
@item -deinterlace
 Deinterlace pictures.
@item -ilme
 Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
 Use this option if your input file is interlaced and you want
 to keep the interlaced format for minimum losses.
 The alternative is to deinterlace the input stream with
@option{-deinterlace}, but deinterlacing introduces losses.
@item -psnr
 Calculate PSNR of compressed frames.
@item -vstats
 Dump video coding statistics to @file{vstats_HHMMSS.log}.
@item -vstats_file @var{file}
 Dump video coding statistics to @var{file}.
@item -top @var{n}
 top=1/bottom=0/auto=-1 field first
@item -dc @var{precision}
 Intra_dc_precision.
@item -vtag @var{fourcc/tag}
 Force video tag/fourcc.
@item -qphist
 Show QP histogram.
@item -vbsf @var{bitstream_filter}
 Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump".
@example
 ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
@end example
@end table
@section Audio Options
@table @option
@item -aframes @var{number}
 Set the number of audio frames to record.
@item -ar @var{freq}
 Set the audio sampling frequency (default = 44100 Hz).
@item -ab @var{bitrate}
 Set the audio bitrate in bit/s (default = 64k).
@item -aq @var{q}
 Set the audio quality (codec-specific, VBR).
@item -ac @var{channels}
 Set the number of audio channels (default = 1).
@item -an
 Disable audio recording.
@item -acodec @var{codec}
 Force audio codec to @var{codec}. Use the @code{copy} special value to
 specify that the raw codec data must be copied as is.
@item -newaudio
 Add a new audio track to the output file. If you want to specify parameters,
 do so before @code{-newaudio} (@code{-acodec}, @code{-ab}, etc..).
 Mapping will be done automatically, if the number of output streams is equal to
 the number of input streams, else it will pick the first one that matches. You
 can override the mapping using @code{-map} as usual.
 Example:
@example
 ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
@end example
@item -alang @var{code}
 Set the ISO 639 language code (3 letters) of the current audio stream.
@end table
@section Advanced Audio options:
@table @option
@item -atag @var{fourcc/tag}
 Force audio tag/fourcc.
@item -absf @var{bitstream_filter}
 Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
@end table
@section Subtitle options:
@table @option
@item -scodec @var{codec}
 Force subtitle codec ('copy' to copy stream).
@item -newsubtitle
 Add a new subtitle stream to the current output stream.
@item -slang @var{code}
 Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -sn
 Disable subtitle recording.
@item -sbsf @var{bitstream_filter}
 Bitstream filters available are "mov2textsub", "text2movsub".
@example
 ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
@end example
@end table
@section Audio/Video grab options
@table @option
@item -vc @var{channel}
 Set video grab channel (DV1394 only).
@item -tvstd @var{standard}
 Set television standard (NTSC, PAL (SECAM)).
@item -isync
 Synchronize read on input.
@end table
@section Advanced options
@table @option
@item -map @var{input_stream_id}[:@var{sync_stream_id}]
 Set stream mapping from input streams to output streams.
 Just enumerate the input streams in the order you want them in the output.
@var{sync_stream_id} if specified sets the input stream to sync
 against.
@item -map_meta_data @var{outfile}:@var{infile}
 Set meta data information of @var{outfile} from @var{infile}.
@item -debug
 Print specific debug info.
@item -benchmark
 Show benchmarking information at the end of an encode.
 Shows CPU time used and maximum memory consumption.
 Maximum memory consumption is not supported on all systems,
 it will usually display as 0 if not supported.
@item -dump
 Dump each input packet.
@item -hex
 When dumping packets, also dump the payload.
@item -bitexact
 Only use bit exact algorithms (for codec testing).
@item -ps @var{size}
 Set RTP payload size in bytes.
@item -re
 Read input at native frame rate. Mainly used to simulate a grab device.
@item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@item -loop_output @var{number_of_times}
 Repeatedly loop output for formats that support looping such as animated GIF
 (0 will loop the output infinitely).
@item -threads @var{count}
 Thread count.
@item -vsync @var{parameter}
 Video sync method.
 0   Each frame is passed with its timestamp from the demuxer to the muxer
 1   Frames will be duplicated and dropped to achieve exactly the requested
    constant framerate.
 2   Frames are passed through with their timestamp or dropped so as to prevent
    2 frames from having the same timestamp
 -1  Chooses between 1 and 2 depending on muxer capabilities. This is the default method.
 With -map you can select from
 which stream the timestamps should be taken. You can leave either video or
 audio unchanged and sync the remaining stream(s) to the unchanged one.
@item -async @var{samples_per_second}
 Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
 the parameter is the maximum samples per second by which the audio is changed.
 -async 1 is a special case where only the start of the audio stream is corrected
 without any later correction.
@item -copyts
 Copy timestamps from input to output.
@item -shortest
 Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
 Timestamp discontinuity delta threshold.
@item -muxdelay @var{seconds}
 Set the maximum demux-decode delay.
@item -muxpreload @var{seconds}
 Set the initial demux-decode delay.
@end table
@section Preset files
 A preset file contains a sequence of @var{option}=@var{value} pairs,
 one for each line, specifying a sequence of options which would be
 awkward to specify on the command line. Lines starting with the hash
 ('#') character are ignored and are used to provide comments. Check
 the @file{ffpresets} directory in the FFmpeg source tree for examples.
 Preset files are specified with the @code{vpre}, @code{apre},
@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
 filename of the preset instead of a preset name as input and can be
 used for any kind of codec. For the @code{vpre}, @code{apre}, and
@code{spre} options, the options specified in a preset file are
 applied to the currently selected codec of the same type as the preset
 option.
 The argument passed to the @code{vpre}, @code{apre}, and @code{spre}
 preset options identifies the preset file to use according to the
 following rules:
 First ffmpeg searches for a file named @var{arg}.ffpreset in the
 directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
 the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg})
 in that order. For example, if the argument is @code{libx264-max}, it will
 search for the file @file{libx264-max.ffpreset}.
 If no such file is found, then ffmpeg will search for a file named
@var{codec_name}-@var{arg}.ffpreset in the above-mentioned
 directories, where @var{codec_name} is the name of the codec to which
 the preset file options will be applied. For example, if you select
 the video codec with @code{-vcodec libx264} and use @code{-vpre max},
 then it will search for the file @file{libx264-max.ffpreset}.
@anchor{FFmpeg formula evaluator}
@section FFmpeg formula evaluator
 When evaluating a rate control string, FFmpeg uses an internal formula
 evaluator.
 The following binary operators are available: @code{+}, @code{-},
@code{*}, @code{/}, @code{^}.
 The following unary operators are available: @code{+}, @code{-},
@code{(...)}.
 The following statements are available: @code{ld}, @code{st},
@code{while}.
 The following functions are available:
@table @var
@item sinh(x)
@item cosh(x)
@item tanh(x)
@item sin(x)
@item cos(x)
@item tan(x)
@item atan(x)
@item asin(x)
@item acos(x)
@item exp(x)
@item log(x)
@item abs(x)
@item squish(x)
@item gauss(x)
@item mod(x, y)
@item max(x, y)
@item min(x, y)
@item eq(x, y)
@item gte(x, y)
@item gt(x, y)
@item lte(x, y)
@item lt(x, y)
@item bits2qp(bits)
@item qp2bits(qp)
@end table
 The following constants are available:
@table @var
@item PI
@item E
@item iTex
@item pTex
@item tex
@item mv
@item fCode
@item iCount
@item mcVar
@item var
@item isI
@item isP
@item isB
@item avgQP
@item qComp
@item avgIITex
@item avgPITex
@item avgPPTex
@item avgBPTex
@item avgTex
@end table
@c man end
@ignore
@setfilename ffmpeg
@settitle FFmpeg video converter
@c man begin SEEALSO
 ffserver(1), ffplay(1) and the HTML documentation of @file{ffmpeg}.
@c man end
@c man begin AUTHOR
 Fabrice Bellard
@c man end
@end ignore
@section Protocols
 The file name can be @file{-} to read from standard input or to write
 to standard output.
 FFmpeg also handles many protocols specified with an URL syntax.
 Use 'ffmpeg -protocols' to see a list of the supported protocols.
 The protocol @code{http:} is currently used only to communicate with
 FFserver (see the FFserver documentation). When FFmpeg will be a
 video player it will also be used for streaming :-)
@chapter Tips
@itemize
@item For streaming at very low bitrate application, use a low frame rate
 and a small GOP size. This is especially true for RealVideo where
 the Linux player does not seem to be very fast, so it can miss
 frames. An example is:
@example
 ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
@end example
@item  The parameter 'q' which is displayed while encoding is the current
 quantizer. The value 1 indicates that a very good quality could
 be achieved. The value 31 indicates the worst quality. If q=31 appears
 too often, it means that the encoder cannot compress enough to meet
 your bitrate. You must either increase the bitrate, decrease the
 frame rate or decrease the frame size.
@item If your computer is not fast enough, you can speed up the
 compression at the expense of the compression ratio. You can use
 '-me zero' to speed up motion estimation, and '-intra' to disable
 motion estimation completely (you have only I-frames, which means it
 is about as good as JPEG compression).
@item To have very low audio bitrates, reduce the sampling frequency
 (down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
@item To have a constant quality (but a variable bitrate), use the option
 '-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
 quality).
@item When converting video files, you can use the '-sameq' option which
 uses the same quality factor in the encoder as in the decoder.
 It allows almost lossless encoding.
@end itemize
@bye
--- a/doc/ffmpeg-filters.texi
+++ b/doc/ffmpeg-filters.texi
@@ -1,43 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Filters Documentation
@titlepage
@center @titlefont{FFmpeg Filters Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 This document describes filters, sources, and sinks provided by the
 libavfilter library.
@c man end DESCRIPTION
@include filters.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavfilter.html,libavfilter}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-filters
@settitle FFmpeg filters
@end ignore
@bye
--- a/doc/ffmpeg-formats.texi
+++ b/doc/ffmpeg-formats.texi
@@ -1,43 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Formats Documentation
@titlepage
@center @titlefont{FFmpeg Formats Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 This document describes the supported formats (muxers and demuxers)
 provided by the libavformat library.
@c man end DESCRIPTION
@include formats.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-formats
@settitle FFmpeg formats
@end ignore
@bye
--- a/doc/ffmpeg-protocols.texi
+++ b/doc/ffmpeg-protocols.texi
@@ -1,43 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Protocols Documentation
@titlepage
@center @titlefont{FFmpeg Protocols Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 This document describes the input and output protocols provided by the
 libavformat library.
@c man end DESCRIPTION
@include protocols.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-protocols
@settitle FFmpeg protocols
@end ignore
@bye
--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -1,45 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Resampler Documentation
@titlepage
@center @titlefont{FFmpeg Resampler Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 The FFmpeg resampler provides a high-level interface to the
 libswresample library audio resampling utilities. In particular it
 allows one to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.
@c man end DESCRIPTION
@include resampler.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswresample.html,libswresample}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-resampler
@settitle FFmpeg Resampler
@end ignore
@bye
--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@@ -1,44 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Scaler Documentation
@titlepage
@center @titlefont{FFmpeg Scaler Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 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
@include scaler.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswscale.html,libswscale}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-scaler
@settitle FFmpeg video scaling and pixel format converter
@end ignore
@bye
--- a/doc/ffmpeg-utils.texi
+++ b/doc/ffmpeg-utils.texi
@@ -1,43 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Utilities Documentation
@titlepage
@center @titlefont{FFmpeg Utilities Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
 This document describes some generic features and utilities provided
 by the libavutil library.
@c man end DESCRIPTION
@include utils.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
 ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-utils
@settitle FFmpeg utilities
@end ignore
@bye
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
--- a/doc/ffmpeg.txt
+++ b/doc/ffmpeg.txt
@@ -1,47 +0,0 @@
                                                                                                                                       :
                                             ffmpeg.c                                                                                  :       libav*
                                             ========                                                                                  :       ======
                                                                                                                                       :
                                                                                                                                       :
                                                                                                       --------------------------------:---> AVStream...
                                                                    InputStream input_streams[]      /                                 :
                                                                                                    /                                  :
                    InputFile input_files[]                         +==========================+   /   ^                               :
                                                          ------> 0 |      : st ---:-----------:--/    :                               :
                 ^  +------+-----------+-----+          /           +--------------------------+       :                               :
                 :  |      :ist_index--:-----:---------/          1 |      : st    :           |       :                               :
                 :  +------+-----------+-----+                      +==========================+       :                               :
 nb_input_files  :  |      :ist_index--:-----:------------------> 2 |      : st    :           |       :                               :
                 :  +------+-----------+-----+                      +--------------------------+       :  nb_input_streams             :
                 :  |      :ist_index  :     |                    3 |            ...           |       :                               :
                 v  +------+-----------+-----+                      +--------------------------+       :                               :
                                                              --> 4 |                          |       :                               :
                                                             |      +--------------------------+       :                               :
                                                             |    5 |                          |       :                               :
                                                             |      +==========================+       v                               :
                                                             |                                                                         :
                                                             |                                                                         :
                                                             |                                                                         :
                                                             |                                                                         :
                                                              ---------                                --------------------------------:---> AVStream...
                                                                        \                            /                                 :
                                                                    OutputStream output_streams[]   /                                  :
                                                                          \                        /                                   :
                                                                    +======\======================/======+      ^                      :
                                                          ------> 0 |   : source_index  : st-:---        |      :                      :
                    OutputFile output_files[]           /           +------------------------------------+      :                      :
                                                       /          1 |   :               :    :           |      :                      :
                 ^  +------+------------+-----+       /             +------------------------------------+      :                      :
                 :  |      : ost_index -:-----:------/            2 |   :               :    :           |      :                      :
 nb_output_files :  +------+------------+-----+                     +====================================+      :                      :
                 :  |      : ost_index -:-----|-----------------> 3 |   :               :    :           |      :                      :
                 :  +------+------------+-----+                     +------------------------------------+      : nb_output_streams    :
                 :  |      :            :     |                   4 |                                    |      :                      :
                 :  +------+------------+-----+                     +------------------------------------+      :                      :
                 :  |      :            :     |                   5 |                                    |      :                      :
                 v  +------+------------+-----+                     +------------------------------------+      :                      :
                                                                  6 |                                    |      :                      :
                                                                    +------------------------------------+      :                      :
                                                                  7 |                                    |      :                      :
                                                                    +====================================+      v                      :
                                                                                                                                       :
--- a/doc/ffmpeg_powerpc_performance_evaluation_howto.txt
+++ b/doc/ffmpeg_powerpc_performance_evaluation_howto.txt
@@ -0,0 +1,172 @@
 FFmpeg & evaluating performance on the PowerPC Architecture HOWTO
 (c) 2003-2004 Romain Dolbeau <romain@dolbeau.org>
 I - Introduction
 The PowerPC architecture and its SIMD extension AltiVec offer some
 interesting tools to evaluate performance and improve the code.
 This document tries to explain how to use those tools with FFmpeg.
 The architecture itself offers two ways to evaluate the performance of
 a given piece of code:
 1) The Time Base Registers (TBL)
 2) The Performance Monitor Counter Registers (PMC)
 The first ones are always available, always active, but they're not very
 accurate: the registers increment by one every four *bus* cycles. On
 my 667 Mhz tiBook (ppc7450), this means once every twenty *processor*
 cycles. So we won't use that.
 The PMC are much more useful: not only can they report cycle-accurate
 timing, but they can also be used to monitor many other parameters,
 such as the number of AltiVec stalls for every kind of instruction,
 or instruction cache misses. The downside is that not all processors
 support the PMC (all G3, all G4 and the 970 do support them), and
 they're inactive by default - you need to activate them with a
 dedicated tool. Also, the number of available PMC depends on the
 procesor: the various 604 have 2, the various 75x (aka. G3) have 4,
 and the various 74xx (aka G4) have 6.
 *WARNING*: The PowerPC 970 is not very well documented, and its PMC
 registers are 64 bits wide. To properly notify the code, you *must*
 tune for the 970 (using --tune=970), or the code will assume 32 bit
 registers.
 II - Enabling FFmpeg PowerPC performance support
 This needs to be done by hand. First, you need to configure FFmpeg as
 usual, but add the "--powerpc-perf-enable" option. For instance:
 #####
 ./configure --prefix=/usr/local/ffmpeg-svn --cc=gcc-3.3 --tune=7450 --powerpc-perf-enable
 #####
 This will configure FFmpeg to install inside /usr/local/ffmpeg-svn,
 compiling with gcc-3.3 (you should try to use this one or a newer
 gcc), and tuning for the PowerPC 7450 (i.e. the newer G4; as a rule of
 thumb, those at 550Mhz and more). It will also enable the PMC.
 You may also edit the file "config.h" to enable the following line:
 #####
 // #define ALTIVEC_USE_REFERENCE_C_CODE 1
 #####
 If you enable this line, then the code will not make use of AltiVec,
 but will use the reference C code instead. This is useful to compare
 performance between two versions of the code.
 Also, the number of enabled PMC is defined in "libavcodec/ppc/dsputil_ppc.h":
 #####
 #define POWERPC_NUM_PMC_ENABLED 4
 #####
 If you have a G4 CPU, you can enable all 6 PMC. DO NOT enable more
 PMC than available on your CPU!
 Then, simply compile FFmpeg as usual (make && make install).
 III - Using FFmpeg PowerPC performance support
 This FFmeg can be used exactly as usual. But before exiting, FFmpeg
 will dump a per-function report that looks like this:
 #####
 PowerPC performance report
 Values are from the PMC registers, and represent whatever the
 registers are set to record.
 Function "gmc1_altivec" (pmc1):
        min: 231
        max: 1339867
        avg: 558.25 (255302)
 Function "gmc1_altivec" (pmc2):
        min: 93
        max: 2164
        avg: 267.31 (255302)
 Function "gmc1_altivec" (pmc3):
        min: 72
        max: 1987
        avg: 276.20 (255302)
 (...)
 #####
 In this example, PMC1 was set to record CPU cycles, PMC2 was set to
 record AltiVec Permute Stall Cycles, and PMC3 was set to record AltiVec
 Issue Stalls.
 The function "gmc1_altivec" was monitored 255302 times, and the
 minimum execution time was 231 processor cycles. The max and average
 aren't much use, as it's very likely the OS interrupted execution for
 reasons of its own :-(
 With the exact same settings and source file, but using the reference C
 code we get:
 #####
 PowerPC performance report
 Values are from the PMC registers, and represent whatever the
 registers are set to record.
 Function "gmc1_altivec" (pmc1):
        min: 592
        max: 2532235
        avg: 962.88 (255302)
 Function "gmc1_altivec" (pmc2):
        min: 0
        max: 33
        avg: 0.00 (255302)
 Function "gmc1_altivec" (pmc3):
        min: 0
        max: 350
        avg: 0.03 (255302)
 (...)
 #####
 592 cycles, so the fastest AltiVec execution is about 2.5x faster than
 the fastest C execution in this example. It's not perfect but it's not
 bad (well I wrote this function so I can't say otherwise :-).
 Once you have that kind of report, you can try to improve things by
 finding what goes wrong and fixing it; in the example above, one
 should try to diminish the number of AltiVec stalls, as this *may*
 improve performance.
 IV) Enabling the PMC in Mac OS X
 This is easy. Use "Monster" and "monster". Those tools come from
 Apple's CHUD package, and can be found hidden in the developer web
 site & FTP site. "MONster" is the graphical application, use it to
 generate a config file specifying what each register should
 monitor. Then use the command-line application "monster" to use that
 config file, and enjoy the results.
 Note that "MONster" can be used for many other things, but it's
 documented by Apple, it's not my subject.
 If you are using CHUD 4.4.2 or later, you'll notice that MONster is
 no longer available. It's been superseeded by Shark, where
 configuration of PMCs is available as a plugin.
 V) Enabling the PMC on Linux
 On linux you may use oprofile from http://oprofile.sf.net, depending on the
 version and the cpu you may need to apply a patch[1] to access a set of the
 possibile counters from the userspace application. You can always define them
 using the kernel interface /dev/oprofile/* .
 [1] http://dev.gentoo.org/~lu_zero/development/oprofile-g4-20060423.patch
 --
 Romain Dolbeau <romain@dolbeau.org>
 Luca Barbato <lu_zero@gentoo.org>
--- a/doc/ffplay-doc.texi
+++ b/doc/ffplay-doc.texi
@@ -0,0 +1,160 @@
 \input texinfo @c -*- texinfo -*-
@settitle FFplay Documentation
@titlepage
@sp 7
@center @titlefont{FFplay Documentation}
@sp 3
@end titlepage
@chapter Introduction
@c man begin DESCRIPTION
 FFplay is a very simple and portable media player using the FFmpeg
 libraries and the SDL library. It is mostly used as a testbed for the
 various FFmpeg APIs.
@c man end
@chapter Invocation
@section Syntax
@example
@c man begin SYNOPSIS
 ffplay [options] @file{input_file}
@c man end
@end example
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -x @var{width}
 Force displayed width.
@item -y @var{height}
 Force displayed height.
@item -s @var{size}
 Set frame size (WxH or abbreviation), needed for videos which don't
 contain a header with the frame size like raw YUV.
@item -an
 Disable audio.
@item -vn
 Disable video.
@item -ss @var{pos}
 Seek to a given position in seconds.
@item -t @var{duration}
 play <duration> seconds of audio/video
@item -bytes
 Seek by bytes.
@item -nodisp
 Disable graphical display.
@item -f @var{fmt}
 Force format.
@item -window_title @var{title}
 Set window title (default is the input filename).
@item -loop @var{number}
 Loops movie playback <number> times. 0 means forever.
@end table
@section Advanced options
@table @option
@item -pix_fmt @var{format}
 Set pixel format.
@item -stats
 Show the stream duration, the codec parameters, the current position in
 the stream and the audio/video synchronisation drift.
@item -debug
 Print specific debug info.
@item -bug
 Work around bugs.
@item -vismv
 Visualize motion vectors.
@item -fast
 Non-spec-compliant optimizations.
@item -genpts
 Generate pts.
@item -rtp_tcp
 Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
 if you are streaming with the RTSP protocol.
@item -sync @var{type}
 Set the master clock to audio (@code{type=audio}), video
 (@code{type=video}) or external (@code{type=ext}). Default is audio. The
 master clock is used to control audio-video synchronization. Most media
 players use audio as master clock, but in some cases (streaming or high
 quality broadcast) it is necessary to change that. This option is mainly
 used for debugging purposes.
@item -threads @var{count}
 Set the thread count.
@item -ast @var{audio_stream_number}
 Select the desired audio stream number, counting from 0. The number
 refers to the list of all the input audio streams. If it is greater
 than the number of audio streams minus one, then the last one is
 selected, if it is negative the audio playback is disabled.
@item -vst @var{video_stream_number}
 Select the desired video stream number, counting from 0. The number
 refers to the list of all the input video streams. If it is greater
 than the number of video streams minus one, then the last one is
 selected, if it is negative the video playback is disabled.
@item -sst @var{subtitle_stream_number}
 Select the desired subtitle stream number, counting from 0. The number
 refers to the list of all the input subtitle streams. If it is greater
 than the number of subtitle streams minus one, then the last one is
 selected, if it is negative the subtitle rendering is disabled.
@end table
@section While playing
@table @key
@item q, ESC
 Quit.
@item f
 Toggle full screen.
@item p, SPC
 Pause.
@item a
 Cycle audio channel.
@item v
 Cycle video channel.
@item t
 Cycle subtitle channel.
@item w
 Show audio waves.
@item left/right
 Seek backward/forward 10 seconds.
@item down/up
 Seek backward/forward 1 minute.
@item mouse click
 Seek to percentage in file corresponding to fraction of width.
@end table
@c man end
@ignore
@setfilename ffplay
@settitle FFplay media player
@c man begin SEEALSO
 ffmpeg(1), ffserver(1) and the HTML documentation of @file{ffmpeg}.
@c man end
@c man begin AUTHOR
 Fabrice Bellard
@c man end
@end ignore
@bye
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -1,302 +0,0 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffplay Documentation
@titlepage
@center @titlefont{ffplay Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
 ffplay [@var{options}] [@file{input_file}]
@chapter Description
@c man begin DESCRIPTION
 FFplay is a very simple and portable media player using the FFmpeg
 libraries and the SDL library. It is mostly used as a testbed for the
 various FFmpeg APIs.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -x @var{width}
 Force displayed width.
@item -y @var{height}
 Force displayed height.
@item -s @var{size}
 Set frame size (WxH or abbreviation), needed for videos which do
 not contain a header with the frame size like raw YUV.  This option
 has been deprecated in favor of private options, try -video_size.
@item -fs
 Start in fullscreen mode.
@item -an
 Disable audio.
@item -vn
 Disable video.
@item -sn
 Disable subtitles.
@item -ss @var{pos}
 Seek to a given position in seconds.
@item -t @var{duration}
 play <duration> seconds of audio/video
@item -bytes
 Seek by bytes.
@item -nodisp
 Disable graphical display.
@item -f @var{fmt}
 Force format.
@item -window_title @var{title}
 Set window title (default is the input filename).
@item -loop @var{number}
 Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
 Set the show mode to use.
 Available values for @var{mode} are:
@table @samp
@item 0, video
 show video
@item 1, waves
 show audio waves
@item 2, rdft
 show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
@end table
 Default value is "video", if video is not present or cannot be played
 "rdft" is automatically selected.
 You can interactively cycle through the available show modes by
 pressing the key @key{w}.
@item -vf @var{filtergraph}
 Create the filtergraph specified by @var{filtergraph} and use it to
 filter the video stream.
@var{filtergraph} is a description of the filtergraph to apply to
 the stream, and must have a single video input and a single video
 output. In the filtergraph, the input is associated to the label
@code{in}, and the output to the label @code{out}. See the
 ffmpeg-filters manual for more information about the filtergraph
 syntax.
 You can specify this parameter multiple times and cycle through the specified
 filtergraphs along with the show modes by pressing the key @key{w}.
@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
@section Advanced options
@table @option
@item -pix_fmt @var{format}
 Set pixel format.
 This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
 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 -fast
 Non-spec-compliant optimizations.
@item -genpts
 Generate pts.
@item -sync @var{type}
 Set the master clock to audio (@code{type=audio}), video
 (@code{type=video}) or external (@code{type=ext}). Default is audio. The
 master clock is used to control audio-video synchronization. Most media
 players use audio as master clock, but in some cases (streaming or high
 quality broadcast) it is necessary to change that. This option is mainly
 used for debugging purposes.
@item -ast @var{audio_stream_specifier}
 Select the desired audio stream using the given stream specifier. The stream
 specifiers are described in the @ref{Stream specifiers} chapter. If this option
 is not specified, the "best" audio stream is selected in the program of the
 already selected video stream.
@item -vst @var{video_stream_specifier}
 Select the desired video stream using the given stream specifier. The stream
 specifiers are described in the @ref{Stream specifiers} chapter. If this option
 is not specified, the "best" video stream is selected.
@item -sst @var{subtitle_stream_specifier}
 Select the desired subtitle stream using the given stream specifier. The stream
 specifiers are described in the @ref{Stream specifiers} chapter. If this option
 is not specified, the "best" subtitle stream is selected in the program of the
 already selected video or audio stream.
@item -autoexit
 Exit when video is done playing.
@item -exitonkeydown
 Exit if any key is pressed.
@item -exitonmousedown
 Exit if any mouse button is pressed.
@item -codec:@var{media_specifier} @var{codec_name}
 Force a specific decoder implementation for the stream identified by
@var{media_specifier}, which can assume the values @code{a} (audio),
@code{v} (video), and @code{s} subtitle.
@item -acodec @var{codec_name}
 Force a specific audio decoder.
@item -vcodec @var{codec_name}
 Force a specific video decoder.
@item -scodec @var{codec_name}
 Force a specific subtitle decoder.
@item -autorotate
 Automatically rotate the video according to file metadata. Enabled by
 default, use @option{-noautorotate} to disable it.
@item -framedrop
 Drop video frames if video is out of sync. Enabled by default if the master
 clock is not set to video. Use this option to enable frame dropping for all
 master clock sources, use @option{-noframedrop} to disable it.
@item -infbuf
 Do not limit the input buffer size, read as much data as possible from the
 input as soon as possible. Enabled by default for realtime streams, where data
 may be dropped if not read in time. Use this option to enable infinite buffers
 for all inputs, use @option{-noinfbuf} to disable it.
@end table
@section While playing
@table @key
@item q, ESC
 Quit.
@item f
 Toggle full screen.
@item p, SPC
 Pause.
@item a
 Cycle audio channel in the current program.
@item v
 Cycle video channel.
@item t
 Cycle subtitle channel in the current program.
@item c
 Cycle program.
@item w
 Cycle video filters or show modes.
@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.
@item down/up
 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
 Seek to percentage in file corresponding to fraction of width.
@end table
@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},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
 ffplay(1),
@end ifset
@ifset config-not-all
 ffplay-all(1),
@end ifset
 ffmpeg(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffplay
@settitle FFplay media player
@end ignore
@bye
--- a/Show More
+++ b/Show More
`@@ -500,3 +500,5 @@ necessary. Here is a sample; alter the names:`
	`Ty Coon, President of Vice`	`Ty Coon, President of Vice`

	`That's all there is to it!`	`That's all there is to it!`
		`@@ -1,2 +0,0 @@`
			`#!/bin/sh`
			`exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"`